ETERNITY V10 System Manual
ETERNITY V10 System Manual
System Manual
ETERNITY
The IP-PBX with Seamless Mobility
and Universal Connectivity
System Manual
Documentation Disclaimer
Matrix Comsec reserves the right to make changes in the design or components of the product as engineering and
manufacturing may warrant. Specifications are subject to change without notice.
This is a general documentation for all models of the product. The product may not support all the features and
facilities described in the documentation.
Information in this documentation may change from time to time. Matrix Comsec reserves the right to revise
information in this publication for any reason without prior notice. Matrix Comsec makes no warranties with respect
to this documentation and disclaims any implied warranties. While every precaution has been taken in the
preparation of this system manual, Matrix Comsec assumes no responsibility for errors or omissions. Neither is any
liability assumed for damages resulting from the use of the information contained herein.
Neither Matrix Comsec nor its affiliates shall be liable to the purchaser of this product or third parties for damages,
losses, costs or expenses incurred by the purchaser or third parties as a result of: accident, misuse or abuse of this
product or unauthorized modifications, repairs or alterations to this product or failure to strictly comply with Matrix
Comsec's operating and maintenance instructions.
Copyright
All rights reserved. No part of this system manual may be copied or reproduced in any form or by any means
without the prior written consent of Matrix Comsec.
Version 10
Release date: June 18, 2011
Contents
Introduction..................................................................................................................................................... 1
Welcome ............................................................................................................................................................. 1
About this System Manual .................................................................................................................................. 1
Table of Contents i
The Digital Key Phone Card ............................................................................................................................ 188
The Two-Wire Trunk Card ............................................................................................................................... 200
The BRI Card .................................................................................................................................................. 205
The T1E1PRI Card .......................................................................................................................................... 215
The Mobile Card .............................................................................................................................................. 221
The E&M Card ................................................................................................................................................ 224
The VoIP Card ................................................................................................................................................ 230
SIP Extensions ................................................................................................................................................ 236
The Voice Mail System Card ........................................................................................................................... 249
Starting Up ETERNITY GE ............................................................................................................................. 251
ii Table of Contents
Features and Facilities ............................................................................................................................... 735
Abbreviated Dialing ......................................................................................................................................... 735
Access Codes ................................................................................................................................................. 750
Account Codes ................................................................................................................................................ 757
Alarms ............................................................................................................................................................. 765
Alternate Number Dialing ................................................................................................................................ 787
Auto Answer .................................................................................................................................................... 794
Auto Call Back (ACB) ...................................................................................................................................... 799
Auto Redial ...................................................................................................................................................... 805
Automated Control Applications ...................................................................................................................... 811
Automatic Number Translation ........................................................................................................................ 825
Background Music (BGM) ............................................................................................................................... 832
Backup-SMDR ................................................................................................................................................ 835
Backup-System Configuration ......................................................................................................................... 840
Backup-System Software ................................................................................................................................ 843
Barge-In .......................................................................................................................................................... 846
BCCH Selection .............................................................................................................................................. 850
Behind the PBX Application ............................................................................................................................ 854
BITE (Built-In Test Equipment) ....................................................................................................................... 858
Building Intercom ............................................................................................................................................ 862
Call Back on Trunk Ports ................................................................................................................................ 864
Call Budget ...................................................................................................................................................... 870
Call Budget on Trunk ...................................................................................................................................... 874
Call Chaining ................................................................................................................................................... 885
Call Cost Calculation (CCC) ............................................................................................................................ 887
Call Cost Display ............................................................................................................................................. 917
Call Duration Control (CDC) ............................................................................................................................ 919
Call Duration Display ....................................................................................................................................... 928
Call Forward .................................................................................................................................................... 929
Call Forward-Remote ...................................................................................................................................... 936
Call Forward-Scheduled .................................................................................................................................. 939
Call Park .......................................................................................................................................................... 945
Call Hold .......................................................................................................................................................... 948
Call Logs ......................................................................................................................................................... 954
Call Pick Up ..................................................................................................................................................... 958
Call Progress Tones ........................................................................................................................................ 966
Call Restriction based on IP Address .............................................................................................................. 977
Call Taping ...................................................................................................................................................... 980
Call Toggle ...................................................................................................................................................... 989
Call Transfer .................................................................................................................................................... 993
Calling Line Identification and Presentation (CLIP) ....................................................................................... 1001
Calling Line Identity Restriction (CLIR) ......................................................................................................... 1008
Cancel All Station Features ........................................................................................................................... 1010
Class of Service (COS) ................................................................................................................................. 1011
CLI Based Routing ........................................................................................................................................ 1021
Clock Synchronization ................................................................................................................................... 1025
Closed User Group (CUG) ............................................................................................................................ 1028
Closed User Group-With Exchange ID ......................................................................................................... 1033
CO Call Waiting ............................................................................................................................................. 1037
Communication Ports .................................................................................................................................... 1038
Conference Dial-In ........................................................................................................................................ 1044
Conference-3 Party ....................................................................................................................................... 1051
Conference-Multiparty ................................................................................................................................... 1055
Conflict Dialing .............................................................................................................................................. 1061
Conversation Recording ................................................................................................................................ 1064
iv Table of Contents
Meet Me Paging ............................................................................................................................................ 1317
Message Wait ............................................................................................................................................... 1319
Mobility Extension ......................................................................................................................................... 1324
Multi-Stage Dialing ........................................................................................................................................ 1331
Music on Hold (MOH) .................................................................................................................................... 1335
Number Lists ................................................................................................................................................. 1337
Mute .............................................................................................................................................................. 1342
OFF-Hook Alert ............................................................................................................................................. 1344
OG Reference Table ..................................................................................................................................... 1346
Outgoing Trunk Bundle ................................................................................................................................. 1348
OG Trunk Bundle Group ............................................................................................................................... 1352
PCAP Trace .................................................................................................................................................. 1359
Paging ........................................................................................................................................................... 1366
Peer-to-Peer Calling ...................................................................................................................................... 1371
PLCC-An Introduction ................................................................................................................................... 1378
Presence ....................................................................................................................................................... 1386
Priority Calls in E&M MFCR2 Signaling ........................................................................................................ 1393
Priority ........................................................................................................................................................... 1397
Privacy .......................................................................................................................................................... 1400
QSIG ............................................................................................................................................................. 1402
Quick Dial ...................................................................................................................................................... 1415
Raid ............................................................................................................................................................... 1417
RCOC (Return Call to Original Caller) .......................................................................................................... 1419
Real Time Clock (RTC) ................................................................................................................................. 1424
Reminder ....................................................................................................................................................... 1430
Remote Programming ................................................................................................................................... 1439
Room Monitor ................................................................................................................................................ 1441
Routing Group ............................................................................................................................................... 1443
Security Alarm and Reporting ....................................................................................................................... 1447
Selective Port Access ................................................................................................................................... 1456
Self Ring Test ................................................................................................................................................ 1458
SIM Card Balance and Recharging ............................................................................................................... 1459
Software Port and Hardware ID .................................................................................................................... 1462
Static Routing Table ...................................................................................................................................... 1467
Station Message Detail Recording ................................................................................................................ 1472
Station Message Detail Recording-Online .................................................................................................... 1473
Station Message Detail Recording-Posting ................................................................................................... 1489
Station Message Detail Recording-Report .................................................................................................... 1531
Station Message Detail Recording-Storage .................................................................................................. 1545
System Activity Log ....................................................................................................................................... 1551
System Activity Log Display .......................................................................................................................... 1557
System Fault Log Display ............................................................................................................................. 1558
System Fault Log .......................................................................................................................................... 1559
System Parameters ....................................................................................................................................... 1565
System Timers and Counts ........................................................................................................................... 1584
System Security ............................................................................................................................................ 1595
T1 Maintenance ............................................................................................................................................ 1600
T1E1 Trunks .................................................................................................................................................. 1608
T1 RBS Parameters ...................................................................................................................................... 1660
Time Tables .................................................................................................................................................. 1671
Time Zone Display ........................................................................................................................................ 1675
Toll Control .................................................................................................................................................... 1676
Trunk Access Group (TAG) ........................................................................................................................... 1692
Trunk Auto Answer ........................................................................................................................................ 1693
Trunk Landing Group (TLG) .......................................................................................................................... 1700
Trunk Reservation ......................................................................................................................................... 1703
Table of Contents v
User Absent/Present ..................................................................................................................................... 1705
User Password .............................................................................................................................................. 1707
Video Call ...................................................................................................................................................... 1709
Virtual Station ................................................................................................................................................ 1710
Voice Help ..................................................................................................................................................... 1711
Voice Mail Integration .................................................................................................................................... 1712
Voice Message Applications ........................................................................................................................ 1716
Walk-In Class of Service ............................................................................................................................... 1726
vi Table of Contents
CHAPTER 1 Introduction
Welcome
Thank you for choosing the Matrix ETERNITY! We hope you will make optimum use of this intelligent, integrated
Voice Switch. Please read this document carefully to get acquainted with the product before installing and operating
it.
Those who require instructions for quick installation and operation of this system, may refer to the ETERNITY
Quick Start and the ETERNITY User Card shipped with the system.
You may also refer the User Cards for the proprietary digital key phone EON shipped with the system, for
instructions on using the features of the ETERNITY.
This is a common documentation for all models of ETERNITY, namely ETERNITY ME, GE and PE and their
variants.
Intended Audience
No one, other than the System Engineer is permitted to make any alterations to the configuration of the
ETERNITY.
• System Administrators, who are persons who will administer the ETERNITY. Generally an operator/
receptionist in an organization, or the staff manning the reception or front desk area of the establishment
are selected as System Administrators.
• Users, persons/organizations who will use the resources of the ETERNITY. They may be executives,
include personnel of small and medium businesses, large enterprises, front desk and service staff of
Hotels/Motels, hospitals, and other commercial and public organizations/institutions.
• Chapter 1: Introduction - gives an overview of this document, its purpose, intended audience,
organization, terms and conventions used to present information and instructions.
• Chapter 2: Know Your ETERNITY - describes the system and its design, application scenarios, the
available models, the interfaces, and the hardware.
• Chapter 3: Installing ETERNITY - gives step-by-step instructions for preparing for and installing the
ETERNITY in general, like setting up the main distribution frames for the wiring, the safety measures for
protecting the system and persons handling the installation and maintenance.
• Chapter 4: Installing ETERNITY ME - provides step-by-step instructions for installing the ETERNITY ME
and its variants, inserting the cards, connecting the cables and powering the system.
• Chapter 5: Installing ETERNITY GE - contains instructions for installing the ETERNITY GE and its
variants, inserting the cards, connecting the cables lines and powering the system.
• Chapter 6: Installing ETERNITY PE - describes the installation of the ETERNITY PE and its variants,
inserting the cards, connecting the cables lines and powering the system.
• Chapter 7: Configuring ETERNITY - contains description of the different tools and options available to
configure ETERNITY. It provides detailed description of how to configure the various extension and trunk
port types - SLT, DKP, ISDN Terminal, SIP Extensions, TWT, Mobile, VoIP-SIP, T1E1PRI, BRI, E&M and
Magneto - supported by ETERNITY.
• Chapter 8: Features and Facilities - describes in detail, each feature and facility offered by the
ETERNITY. This includes a description of the feature/facility, how it works, and how to program the feature/
facility.
The feature description is arranged alphabetically by Feature Name to make it easy for you to locate the
description you want to look up.
This System Manual is presented in a manner that will help you find the information you need easily and quickly.
Instructions
The instructions in this document are written in a numbered, step-by-step format, as follows. Each step, its outcome
and indication/notification, wherever they occur, have been described.
Access Codes
Access codes are strings of digits dialed by an extension to
The Access Codes provided in the instructions throughout this document, are default access codes. It is possible to
change the Access Codes according to user requirement and preferences. Verify with the Installer/System
Engineer, if the default Access Codes have been changed, and use the codes programmed by the System
Engineer. For more information, read the topic “Access Codes” in this document.
Notices
The following symbols have been used for notices to draw your attention to important items.
Important: to indicate something that requires your special attention or to remind you of
something you might need to do when you are using the system.
Caution: to indicate an action or condition that is likely to result in malfunction or damage to the
system or your property.
Warning: to indicate a hazard or an action that will cause damage to the system and or cause
bodily harm to the user.
Tip: to indicate a helpful hint giving you an alternative way to operate the system or carry out a
procedure, or use a feature more efficiently.
Acronyms have been defined in the text and a list of the same is appended.
The words 'ETERNITY', 'System', 'PBX' are used interchangeably and synonymously to mean all models of
ETERNITY. Wherever necessary, the model and variant names are used.
• CO Lines: The lines subscribed from the CO Network. These may be Two-wire Trunk Lines, ISDN BRI,
ISDN PRI, etc.
• Digital Key Phone (DKP): refers to EON, the proprietary digital key phone of Matrix supplied with the
ETERNITY. The term 'Digital Key Phone' refers to all models of EON.
• Enterprise Application/Features: pertaining to the general and special telephone and call management
features required by business establishments, public and private organizations.
• Extension: it is the port of the PBX to which a telephone instrument (DKP/SLT/ISDN) is connected.
• External Calls: calls made by users of ETERNITY to subscribers of PSTN, PLMN, ITSPs, etc.
• External Numbers: numbers of parties/individuals outside the PBX or PBX network. The unique number
string given to subscribers of PSTN, PLMN, ITSP, etc.
• Hospitality Application/ Features: pertaining to the special telephone and guest/patient management
features required by accommodation establishments like hotels and hospitals.
• Internal Calls: calls made from and received by one extension to another extension of the ETERNITY.
• Mobile Extension: A mobile/landline phone used as a remote extension of ETERNITY. You can access all
the features of an extension of ETERNITY from the mobile/landline phone.
• Port: the physical interfaces on the cards for trunk lines and stations lines.
• Service Provider: the providers of telecom network lines/Internet - POTS, PSTN, GSM, ISDN PRI, ISDN
BRI, and Internet Telephony Service Providers (ITSP).
• Single Line Telephone (SLT): any standard two-wire telephone attached as extensions of the ETERNITY.
• System Administrator Commands/SA Commands: number strings dialed from the System
Administrator access/mode to operate features or set/cancel features for other extensions.
• System Commands/SE Commands: number strings dialed from the System Engineer access/mode to
program the system features/functions.
• TWT trunks: Two-wire trunks, that is, analog trunk lines from the POTS network.
If you encounter any technical problems, please contact your Dealer/reseller or the Matrix Support team.
Introduction
The Matrix ETERNITY is an Integrated Enterprise Voice Switch expandable up to 516 user ports. It is a unique
convergence of innovative switching technology and intelligent software features.
The system is built on PCM/TDM, 100 percent non-blocking, digital technology, providing high density switching. It
is powered by a 32-bit RISC processor for distributed processing. Thus the system offers reliable, efficient, and
unrestricted simultaneous communication (incoming and outgoing) by all users.
Universal Connectivity
ETERNITY offers Universal Connectivity, working with all major telecom interfaces: POTS, ISDN BRI/PRI, T1/E1,
GSM/3G, VoIP, E&M, and Magneto. So, you have access to multiple telecom networks on a single platform. The
system's intelligent Least Cost Routing logic diverts your calls through the appropriate network, ensuring least
possible call cost.
Besides Operator consoles (Digital Key phones, Direct Station Selection consoles) and standard telephones, you
can interface ETERNITY with different types of external devices such as a Fax machine, an external music source,
a public address system.
You can operate security devices with the ETERNITY. Any sensor device such as a smoke detector, an object
sensor, a glass break detector, can be connected to the ETERNITY to instigate a hooter, siren connected to the
ETERNITY.
You can also operate several automated control applications such as a door lock, lights glow signboards, bells,
water pump, and the like.
ETERNITY supports video conferencing and data connectivity. You can interface any standard video phone
conference unit with suitable ISDN Interface with ETERNITY ISDN T1E1PRI and BRI cards.
ETERNITY supports Q-Sig. allowing you to network ETERNITY with another PBX/ETERNITY. So, you have feature
transparency and a network of PBXs working as a single unit.
Intelligent features like Auto Attendant, CLI based Routing and Dial by Name ensure efficient call management and
prompt response to callers.
Least Cost Routing and Call Budgeting help reduce communication cost and enhance productivity.
ETERNITY can route a VoIP call to GSM or T1E1PRI port. In the same way, a call on a T1E1PRI port can be
routed to VoIP, GSM or TWT ports. Further, you can select Fixed or Least Cost Routing to route outgoing calls.
ETERNITY can handle calls on all ports simultaneously, allowing full traffic on all ports.
ETERNITY can work as an adjunct to your existing telephony infrastructure, as a Gateway, saving you the cost of
equipment replacement, wiring and installation, while giving you Universal Connectivity and a host of intelligent
features.
Redundancy
To reduce down time and provide uninterrupted communication, the ETERNITY ME supports redundancy option in
the ETERNITY ME10SR variant for the three cards that are critical to its functioning: The Power Supply Card1, the
Master Card and the Switch Card. There two cards of each on the ETERNITY ME10SR. When the active card fails,
the standby card takes over.
Hot Swap
With the Hot Swap feature2 you can remove a card and insert it back without switching off the system. So, you can
replace a faulty card with a functional one without affecting the functioning of the system.
Q-Sig
With Q-Sig. you can network ETERNITY with another ETERNITY or any other ISDN-PBX to expand the PBX
resources. You can enjoy feature transparency between the PBXs.
Key Features
• Account Codes
• Auto Attendant
• Automatic Call Distribution
• CAS Interface
• Class of Service
• CLI Based Routing
• Closed User Group
1. Redundancy is supported only for the PS48V Power Supply card! It is not supported for the PS UNI Power Supply Card. Refer the
topic “The Power Supply Card” to know more. ETERNITY GE and PE do not support Redundancy.
2. The Hot Swap feature is supported for all cards except Power Supply card in both ETERNITY ME10S and ME16S. The ETERNITY
GE and PE variants do not support Hot Swap.
Also refer “Appendix” for a complete list of Hardware and Software features and technical specifications.
The built-in web server Jeeves allows you to configure the system parameters and features on-site and also from a
remote location using a Web browser.
The Matrix ETERNITY can be deployed in small to large enterprises and institutions: manufacturing units,
corporate offices, banking and financial institutions, software firms, shopping malls, hospitals, hotels-motels, in
power line carrier communication of electric utilities, call centers, in institutions and, power line carrier
communication PLCC networks, as group PBX (GPAX).
The ETERNITY can work as a Gateway with the existing telephony infrastructure - TDM PBX, IP PBX - as
Universal Gateway for Calling Card Operators, Internet Telephony Service Providers (ITSP).
Illustrated in the following are various scenarios where the ETERNITY finds application.
Enterprise Application
The Matrix ETERNITY in available in three models: ETERNITY ME, ETERNITY GE and ETERNITY PE. Each
model has variants with different configurations.
ETERNITY ME
The ETERNITY ME is designed for medium and large organizations and it available in the following variants:
ETERNITY GE
ETERNITY GE is designed for small and medium organizations. It is available in the following variants:
The ETERNITY PE is designed for small and growing organizations/small organizations with growth potential. It
available in the following variants:
ETERNITY PE3SS
ETERNITY PE3SP ETERNITY PE6SP
• 3 universal slots
• 3 universal slots • 6 universal slots
• 24 user ports
• 24 user ports • 48 user ports
ETERNITY ME
The Enclosure
The enclosure of ETERNITY ME consists of 'fixed' and universal slots. The fixed slots are occupied by specific
cards - Power Supply Card, Master Card, Switch Card - and cannot be changed, whereas in the universal slots,
you can install any of the various card.
The slot connectors are located on the motherboard on the backplane of the enclosure. Each slot has guide rails for
inserting the cards.
Illustrated below are design of the enclosure and the position of the slots on each model of ETERNITY ME.
ETERNITY ME16S
In the ETERNITY ME16S, the extreme left slot is reserved for the Power Supply card, the extreme right is reserved
for the Master card, and the second last slot is reserved for the Switch Card. The slots between these fixed slots
are the 16 universal slots to fit the other cards.
ETERNITY ME10S
The ETERNITY ME10SR, which offers the redundancy option, has the same organization of the fixed and universal
slots as the ME10S variant, starting with the Power Supply Card on the extreme left. Only the number of slots
exceeds because of the presence of the second Power Supply Card, Master Card and Switch Card, provided in this
variant to support the Redundancy feature.
The Cards
ETERNITY ME houses the following Cards:
1. Master Card
2. Switch Card
3. Power Supply Cards: PSUNI or PS48V
4. SLT Card
5. TWT+SLT Card
6. DKP
7. Intercom Line Card
8. E&M Card
9. VMS Card
10. BRI Card
11. T1E1PRI Card
12. GSM/3G Card
13. VoIP Card
14. Magneto Card
15. SLT8+MAG2+TWT2+LD2+ENM2 Card
16. SLT8-Magneto8 Card
17. TWT8-Magneto8 Card
The Enclosure
The enclosure of ETERNITY GE has 'fixed' and universal slots. The fixed slots are occupied by specific cards -
Power Card and the CPU Card - and cannot be changed, whereas in the universal slots you can install any of the
various cards.
Inside the enclosure of ETERNITY GE are slot connectors located on the motherboard on the backplane of the
enclosure. Each slot has guide rails for inserting the cards.
Illustrated below are the design of the enclosures and the position of the slots in each model of ETERNITY GE.
ETERNITY GE12S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
ETERNITY GE6S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
ETERNITY GE3S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
1. Power Supply Card - PSUNI Card (90-265V, 47-63Hz Mains as Input AC voltage power supply).
2. CPU Card
3. SLT Card
4. TWT+SLT Card
5. DKP Card
6. DKP+SLT Card
7. Intercom Line Card
8. E&M Card
9. VMS Card
10. BRI Card
11. T1E1PRI Card
12. GSM/3G Card
13. VoIP Card
ETERNITY PE
The Enclosure
The ETERNITY PE has a different design. The enclosure of ETERNITY PE consists of a top plate, which functions
as the cover and can be removed.
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
The CPU card is fixed on the bottom plate of the enclosure, with the 2-row connectors for the card slots facing up.
Illustrated in the following are the design of the enclosures and the position of the slots in each model of ETERNITY
PE.
ETERNITY PE6SP
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
Universal slots are located on the CPU. The connectors of the slots are located on the CPU.
ETERNITY PE3SP
ETERNITY PE3SP is similar to PE6S, except has it has only 3 universal slots.
ETERNITY PE3SS
There is no Communication (COM) Port, no USB port, no Analog Input Port and no Analog Output port.
The Cards
The ETERNITY PE houses the following Cards:
1. SLT Card
2. DKP Card
3. TWT Card
4. DKP+ SLT Card
5. TWT + SLT Card
6. DKP + TWT Card
7. DKP + TWT + SLT Card
8. BRI Card (only ETERNITY PE6SP and PE3SP)
9. T1E1PRI Card (only ETERNITY PE6SP and PE3SP)
10. GSM/3G Card
11. VoIP Card
Card Name ME16S ME10S GE12S GE6S GE3S PE6SP PE3SP PE3SS
Master Card -- -- -- -- -- --
Switch Card -- -- -- -- -- --
CPU Card -- -- ## ## ##
SLT Card
TWT Card -- -- --
TWT+SLT Card
DKP Card
DKP+SLT Card -- --
DKP+TWT Card -- -- -- -- --
DKP+TWT+SLT Card -- -- -- -- --
E&M Card -- -- --
VMS Card
BRI Card --
TIE1PRI Card --
GSM/3G Card
VoIP Card
Magneto Card -- -- -- -- -- --
SLT-Magneto Card -- -- -- -- -- --
TWT-Magneto Card -- -- -- -- -- --
SLT+MAG+TWT2+LD2+ENM2 Card -- -- -- -- -- --
The TWT, SLT, DKP, E&M, VMS, T1E1PRI, BRI, GSM/3G, VoIP, Magneto cards are available in different
configurations for the different models of ETERNITY. For an at-a-glance view of the configurations in which these
cards are available for ETERNITY ME, GE and PE, refer “Technical Specifications”.
The ETERNITY supports the following interfaces for connecting to different telecom networks, digital key phones,
standard telephones and other external devices.
ETERNITY's versatile architecture allows it to be connected to such networks differing in their characteristics. The
TWT Interface supports following features:
• PRI
• Robbed Bit Signaling (RBS)
• Q-Signaling (QSIG)
• E&M
• PRI
• Channel Associated Signaling (CAS)
• Q-Signaling (QSIG)
• E&M
3. T1 PRI (T-Carrier) offers 23 Bearer Channels and one Signaling Channel (23B+D). It is used in North America, Japan and Korea.
4. E1 PRI (E-Carrier) offers 30 Bearer Channels and two Signaling Channels (30B+D). It is used in all countries, except North
America, Japan and Korea.
ISDN BRI
The ISDN BRI Interface enables ETERNITY to be connected to ISDN BRI Lines and connect ISDN BRI compatible
devices with the ETERNITY.
Depending on the requirement, each BRI Port can be configured in the TE/NT mode.
It is possible to feed power from the ETERNITY to the terminal equipment connected to the ETERNITY (on its BRI
port configured as NT).
ETERNITY's Mobile Interface supports full Quad-Band Operation (GSM850, 900, 1800, 1900MHz) for world-wide
use, for Global, Inter and Intra country roaming.
The ETERNITY Mobile Interface does not support GPRS features, Fax and Data services, and network
supported services, except CLIR and USSD.
The VoIP Interface supports Session Initiation Protocol (SIP), the industry standard VoIP.
With SIP Trunks users can make IP calls using the SIP Server of the Internet Telephony Service Providers (ITSPs).
The VoIP Card has an in-built Registrar Server that allows any SIP enabled device like a Wi-Fi mobile handset, a
PDA or an IP-Phone to be registered with it and function as the 'SIP Extension' of the ETERNITY. The SIP
Extension users can make and receive calls to any extension user of the ETERNITY as well as any external
numbers over PSTN, GSM, VoIP and E&M. With SIP Extensions, organizations can communicate and stay
connected at the lowest cost without any geographical restrictions.
The VoIP Interface supports adaptive jitter buffer for reducing delay and improving speech quality.
• STUN.
• VLAN.
• Broad Voice Codec Selection: G.723, G.729ab, GSM FR, iLBC - 30 ms, iLBC - 20 ms, G. 711 µ-Law, and
G. 711 A-Law.
Often, E&M connectivity is used to expand the PBX capacity (by connecting a second PBX with the main PBX) or to
connect two or more remotely located PBXs, forming a network of PBXs.
• Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer “PLCC-An Introduction” to know more.
• Closed User Groups, where several PBXs are connected with each other through E&M tie lines8.
• PBX expansion, where two PBXs are connected with each other with E&M tie lines.
Also, refer the topics “E&M Connectivity” and “E&M Feature Template” to know more.
The E&M Interface can be programmed to provide Trunk Interface, a Subscriber (Station) Interface or both, as a
Tie Line with the dual personality of a Trunk and a Subscriber.
8. The PBXs in a Closed User Group can be connected over ISDN T1/E1 Lines as well. Refer the topic “Closed User Group (CUG)”
to know more.
9. The number of wires used to transmit audio signals.
ETERNITY can land calls from magneto field telephones on the extensions (SLT, DKP, ISDN Terminal) of the
ETERNITY and place calls from the extensions of the ETERNITY on magneto telephones.
To know more about how the Magneto Card works, refer the topic “Configuring Magneto Interface”.
10. This is the line protocol that defines how the equipment seizes the E&M trunk. Also, referred to as Start Dial Supervision Signaling
Protocol.
11. A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator. The
hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is turned to produce
energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the mechanical motion
via the crank to produce sufficient energy to ring other phones or to signal the CO.
The key auto attendant and voicemail features supported by ETERNITY's Voicemail System are:
ETERNITY's Voicemail System also forms the basis of other features like:
• Conversation Recording
• Call Taping
• Voice-guided Wake-up Calls and Reminders
• Message Wait Notification
• Call Transfer to Mailbox
• Call Forward to Voice Mail
• Department Calls - Mailbox for Department Groups
You can also connect ETERNITY to a standalone computer or to a LAN Switch over the Ethernet Port of the
ETERNITY.
The Analog Input Port can handle unamplified, isolated, analog speech signal from an external music source.
Music from this external source can be played as Music-On-Hold to callers and as Background Music to extension
users.
Security Devices
Any type of sensor device like glass break sensor, smoke detector, object sensor, etc. can be connected to the
Digital Input Port of the ETERNITY.
You can connect a Siren or a Hooter to the Digital Output Port of the ETERNITY which can be activated to indicate
emergencies.
The sensor device connected to the Digital Input Port can be used to instigate the hooter or siren connected to the
Digital Output Port.
What's more, these devices can also work on instigation from a sensor connected to the Digital Input Port of the
ETERNITY and can be operated from a remote location using the Direct Inward System Access (DISA) provided by
ETERNITY.
Door Phone
You can connect any standard 4-wire door phones to the Door Phone ports of the ETERNITY. The door phones
can be operated in conjunction with a Door Lock connected to the Digital Output Port of the ETERNITY.
For an at-a-glance view of the maximum trunks and ports available for each of the aforementioned Interface
options on the various models of ETERNITY, refer System Capacity and Resources in Appendix “Technical
Specifications”.
The number of extensions (stations) you require and their location will determine the type of cabling you require on
your premises.
We recommend that you plan the wiring and the installation of the ETERNITY according to your current and
expected future requirements.
Before you begin to install and set up the hardware of ETERNITY, make sure you have the following items:
In simple form, the MDF is a special metallic frame designed and constructed with columns of receptacles to firmly
hold the termination modules for the trunk and extension cables.
A multi-core cable runs from the PBX into the MDF. From the distribution frame, the smaller cables run into each
individual extension telephone outlet or socket (RJ11 or RJ45).
In a multi-storied building or on a widely spread out premises, it is common to have more than one distribution
frame, called the intermediate distribution frame (IDF) on each floor, to provide the connection between the MDF
and the individual telephone wiring. IDFs function as wiring points to gather and distribute wiring. IDFs are useful
used when a large number of stations are to be connected and the wire runs extend over hundreds of feet; hence
the distance is too great to economically terminate every station run individually to the MDF.
• Select a suitable MDF (and IDF, if required) with the standard lead-in cable termination KRONE modules.
• Ensure that the MDF complies with the local building telecom wiring Guidelines, Rules and Regulations.
• This also applies to MDF installed outside the building. It must be protected from exposure to weather
conditions, dust, dampness and humidity
• In washing or toilet facilities, boiler/plant/machine rooms or any area subject to corrosive fumes and
fluids;
• In fire escape stairways;
• Within a cupboard containing a fire hose reel;
• Within any refrigeration room or sauna heater room;
• Near any water feature or water body like fountains, sprinklers, a bath, shower or other fixed water
container, a swimming pool, paddling pool, spa pool or tub; or any area where hosing down operations
are carried out.
• In a high voltage electrical switch room or near a heavy voltage transformer.
• The MDF should be robust and securely attached to a permanent building element such as a wall, floor or
column. Do not mount the MDF on movable elements such as hinged panels or wheeled trolleys.
• Provide adequate space around the MDF where any person is required to pass to enable safe and
convenient access to the MDF and ready escape from the vicinity under emergency conditions.
• Any room containing the MDF must not require the use of a tool, key, card, number pad or the like to exit
the room. Ensure a quick hurdle-free exit from such a room.
• The MDF or the enclosure in which it is located should have the provision for securing with a key, lock or
tool. External MDF should be adequately secured against vandalism and access by children or
unauthorized persons.
Select an appropriate site to install the ETERNITY taking into consideration the following recommendations and
precautions:
• The site of installation should be well-ventilated, moisture and dust free, and not exposed to direct sunlight,
heat, excessive cold or humidity.
• The site should be equidistant from all the extensions to simplify cabling network and reduce cabling costs.
• The system should be installed at a height of at least 3.5 feet from the ground. Installation at this height
makes preventive or corrective maintenance tasks easy.
• The system should be installed away from any source of electromagnetic noise such as any radio
equipment, heavy transformers, faulty electric chokes of tube-lights, any device having faulty coil, etc.
Selecting Cables
• Select standard good quality telephone cables with 0.5 mm conductor diameter for the internal as well as
over-head cabling.
• The length of the cables must not be too long. They must have minimum number of joints. This will help
you detect cable faults easily.
• Maintain cable records so that cables and cross-connections on the MDF can be correctly identified and
connected. The records should be in a clear, legible and updateable format.
• any of the models of the proprietary Digital Key Phone (DKP) of the EON series (EON42, EON48)
You are recommended to connect DKPs with DSS of the EON series for Operator/Receptionist/ Front
Desk/Senior management extensions (stations).
• Arrange for a separate power point and switch, close to the system.
• Power supply for the system must be separate from other heavy electrical loads like Air-conditioners,
heaters, welding machines, electrical motors, etc.
• Terminate all the station cables (connected to the wall sockets/outlets) into the 'Station Lines' side of the
MDF using the punch tool for Krone modules.
• Label the trunk and station line cables for easy identification and keep a record of the trunk and station
lines in an updatable format.
Where multiple wiring cabinets/distribution frames are used, label each frame and reference its number on
the corresponding outlet.
• Install Primary Protection modules with Gas Discharge Tubes (GDT) and fuses on entry points for all trunk
lines. This is to protect the system from heavy voltages from trunk lines and overhead stations.
The product warranty does not cover damages resulting from lack of primary protection on trunk lines.
• It is recommended that you also install Primary Protection modules with GDT and fuses on all Station lines,
particularly off-premise extensions, and E&M ports.
For this, you are recommended to use the Primary Protection Module (PPM4) supplied by Matrix.
• If you are using a smaller configuration of the ETERNITY, like ETERNITY PE3SS, 3SP or 6SP, you may
refer the following diagram to connect the PBX and the Distribution Frame.
You are recommended to use the “Primary Protection Module - PPM4” supplied by Matrix.
The protection can be in the form of surge suppressor devices like Gas Discharge Tubes (GDT), MOVs, Fuses, etc.
The Gas Discharge Tube is an over voltage protection device. It has three terminals. It is connected parallel to the
CO Line or the overhead station cable. The third terminal is connected to a telecom earth. When the voltage
between any of the two terminals exceeds the permissible limit (general 150V), the gas in the device begins to
conduct and the terminals with the earth terminal. Heavy voltage passes to the earth instead of entering the
system, thereby protecting the system.
The Fuses in the PPM4 are an over current protection device. Whenever the current builds up beyond the
permissible limit, (generally 100mA), the fuse opens to protect the circuit ahead.
PPM4 must be properly earthed to work well. It is recommended that PPM4 be connected to a separate telecom
earth (ground).
Telecom earth is a dedicated earth (ground) only for the PBX. A dedicated earth greatly reduces the risk of back
voltage.
Installing PPM4
Refer the block diagram above for the location of the PPM4.
2. Select an appropriate location for the PPM4. Refer the block diagram above when deciding where to place
the PPM4. Also, take into consideration the length of the cables of the PPM4.
3. Use the Mounting Template supplied with the PPM4 to drill holes on the wall to fix the PPM in the selected
location. Fix the screws supplied with the PPM4 into the drilled holes, with their heads protruding from the
wall.
5. To connect cables, press the snap fits on both sides of the PPM4 to release the cover. Remove the cover.
7. Now connect the TWT Trunk wires from the CO side into the PPM4 port connectors marked as P1, P2, P3
and P4.
8. To do so, strip off about half a centimeter of the insulation of the wire ends of the first pair of TWT Trunk
you want to connect to the PPM4.
9. Push back the (orange-color) levers of the connector of port P1, using a blunt pin or a small flat screw
driver or your thumbnail.
10. Insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one wire
in each opening.
11. Release pressure on the levers. Both wires will be held in place by spring clamp action.
12. Now, repeat the above steps to connect the other TWT Trunk wires from the CO side into the connectors of
the ports P2, P3, and P4.
13. Now, terminate the wire pairs emerging from the PPM4 multi-pair cable into the 'Trunk Lines' side of the
MDF using the punch tool for Krone Modules.
14. Replace the cover of the PPM4 by pressing back the snap fits on both sides.
The ETERNITY is an electronic device. When you handle any electrical or electronic equipment, you are in a
situation that could cause you bodily harm, besides damage to the product. When handling any electronic
equipment, you must be aware of the safety hazards involved in electrical circuitry and the standard practices for
accident prevention.
When using any telephone equipment, take every safety precaution to reduce the risk of fire, electric shock and
injury to persons. Read and understand the precautions, dos and don'ts of handling this product listed below.
These instructions are by no means exhaustive. So, take all the necessary precautions for handling electronic and
electrical appliances. Your safety and that of the others lies in your hands.
Location
• Do not place this product in any of the following locations:
• near a water source like a wash bowl, kitchen sink, laundry tub, near a swimming pool, or in a wet
basement.
• In places where dust, oil, corrosive fumes may come in contact with the system.
• Any area where it is exposed to direct sunlight, heat, excessive cold or humidity.
• On moveable or unstable surfaces, which may cause the product to fall and get damaged.
• Any area where shocks or vibration are frequent or strong.
• Near High-Frequency generating devices such as Electric Welder, Sewing Machine or and Microwave
Oven.
• Do not leave cables exposed on the ground where they may be trampled upon, or get damaged by
entangling with feet or pressure from other heavy objects.
Power Supply
• This product should be operated with proper supply voltage. If you are not sure about supply voltage,
contact authorized dealer.
• The ETERNITY does not work in isolation from the environment. Power is fed to the system for functioning
of the system. Being a PBX system, it has several interfaces like trunk lines and extensions, external
music, Public Address System, Printer interface, PC interface, etc. So there are chances of heavy voltages
entering the system through these interfaces. Also, static charges could find their way through the system
components.
• Heavy voltage line falling on the CO line or on the overhead stations cable. A dangerous surge can
occur if a telephone line comes in contact with a power line.
• Lightning/Thunderbolts.
• To protect ETERNITY from these voltages, use Primary Protection/Surge Protectors on the trunk and long
distance extension lines to protect the system from lightning and electrical surges.
• Install any standard Input Protection (punch down protection) on the Krone Modules of the MDF or the
“Primary Protection Module - PPM4” supplied by Matrix at entry points for all CO trunks lines and all
overhead stations. The product Warranty does not cover damages resulting from heavy voltage on CO
lines and overhead stations!
• It is recommended that you install the PPM on the MDF, as MDF cables from the CO are terminated on the
System MDF.
• Generally, persons installing or handling electronic and electrical equipment take precaution to wear
appropriate footwear to get protection from electric shocks. Doing so, the static charge accumulates in his/
her body and does not find its way to the ground. But when such a person touches any of the electronic
cards, the static charge finds its way through the electronic components thereby causing damage to the
cards.
• So, the person installing or servicing the system must provide a path to the static charges, by wearing an
antistatic belt, which is properly earthed.
• If an electrical wire carrying heavy voltage accidentally shorts with this cable, heavy voltages can damage
the communication port.
• It is recommended that the communication cable (connecting ETERNITY and the PC) be run through the
conduit carrying telephone cables or through a separate conduit.
• Heavy voltages on the cable connecting the system and the external music source due to shorting with any
electrical wire.
• An audio signal, not complying with the specifications of this port, is fed to this port.
• Refer the technical specifications of the AIP before connecting any external music device to it.
Protecting the system from heavy voltage on the Analog Output Port
The Analog Output Port of the ETERNITY should be protected from:
• Heavy voltages on the cable connecting the system and the amplifier/speaker due to shorting with any
electrical wire.
Protecting the system from heavy voltage on the Digital Output Port
The Digital Output Port (DOP) of the system should be protected from:
• Heavy voltages on the cable connecting the system and the device connected to the DOP.
Protecting the system from heavy voltage on the Digital Input Port
The Digital Input Port (DIP) of the system should be protected from:
• Heavy voltages on the cable connecting the system and the sensor device or panic switch connected to
the DIP.
• Provide a separate Telecom Earth (Ground) to the system installation. Providing a separate earth to the
telecom equipment eliminates the possibility of any back-voltage on the earth.
• Refer “How to Make the Telecom Earth” for instructions on making the perfect earth (ground).
• Slots and openings in the cabinet and the back or bottom are provided for ventilation, to protect the system
from overheating. These openings must not be blocked or covered.
• Never insert or push objects of any kind into this product through the cabinet slots as they may touch
dangerous voltage points or short out parts which may result in fire or electric shock.
• Do not allow anything to rest on the power cord. Do not locate this product where the cord will be trampled
upon or get entangled.
• This product is equipped with a plug having a third (ground) pin, which fits only into a grounding-type
outlet. This is a safety feature. So, if the existing outlet is not a three-pin and or if you are unable to insert
the plug into the outlet, have the outlet replaced by the electrician.
• Do not overload wall outlets and extension cords as this can result in the risk of fire or electric shock.
• Do not disassemble this product. Opening or removing covers may expose you to dangerous voltages or
other risks. Incorrect reassembly may cause electric shock when the appliance is used. Take the product
to a qualified technician when service or repair work is required.
• Avoid using a telephone (other than a cordless type) during a storm, to prevent electric shock from
lightning.
• Do not use the telephone to report a gas leak in the vicinity of the leak so as to prevent the risk of fire.
External Devices
• When you connect external devices like headset, external music source (PC, Cassette Player, CD Player),
relay devices (door lock, door lock release), sensors, public address or paging devices, telephone
instruments, cables, connectors, etc., ensure that they are of standard make and good quality, so that the
functioning of the system is not affected.
• Matrix does not guarantee the performance of external devices that are not supplied by it.
• Do not use liquid cleaners or aerosol cleaners. Never spill liquid of any kind on the product.
Disposal
• This product must be disposed according to the national laws and regulations prevailing in the country
where it is installed.
• Make sure that the RF Antenna is installed at least 20 cm away from other electronic and radio
transmission devices.
• Make sure that the RF Antenna is installed at a place at 20 cm away from people's vicinity.
• People carrying medical implants like cardiac pacemakers are advised to maintain appropriate distance
from the system. They are also advised to avoid being in the vicinity of the product for a long time.
• The Matrix ETERNITY is to be installed by persons trained and experienced in telecom wiring.
• The person installing the ETERNITY must be familiar with trunks, physical wiring of the MDF on both
the exchange (PBX) side and the line side (CO).
• When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
system and its cards to prevent damage to the system and harm to yourself.
• Use a grounding mat and wear an anti-static strap/belt. Read the dos and don'ts listed in '“Protecting
ETERNITY and Yourself”.
• If you have complied with the requirements and instructions described in “Before You Start”, you may
now begin the installation of your ETERNITY ME.
The Matrix ETERNITY ME is shipped factory fitted with the Power supply card, the Master and Switch Card in their
respective fixed slots (refer the section “Know Your ETERNITY”).
The cards - BRI, T1E1PRI, GSM, VoIP, DKP, TWT, SLT, VMS, E&M, Magneto - are shipped separately as per the
order placed by individual customers. These cards can be installed in any of the Universal slots.
Illustrated below is the position of the fixed and universal slots in each variant of ETERNITY ME.
ETERNITY ME16S
In the ETERNITY ME16S, the extreme left slot is reserved for the Power Supply card, the extreme right is reserved
for the Master card, and the second last slot is reserved for the Switch Card. The slots between these fixed slots
are the 16 universal slots to fit the other cards.
In the ETERNITY ME10S, the first three slots from extreme left slot are reserved for the Power Supply card, the
Master Card and the Switch Card respectively. The remaining slots are the 10 universal slots.
The ETERNITY ME10SR, which offers the redundancy option, has the same organization of the fixed and universal
slots as the ME10S variant, starting with the Power Supply Card on the extreme left. Only, the number of slots
exceeds on account of the second Power Supply Card, Master Card, and Switch Card, provided in this variant to
support the Redundancy feature.
Follow the installation instructions for cards described here, also when you expand the system (add more cards) or
remove or swap cards for maintenance and repair.
1. Unpack the box. Check the package contents (see “Packing List”). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation.
4. Mount the system at the selected site. Make sure that the system is place such that you have full access to
the front and back panels. The holes in the panels are provided for ventilation; Make sure that these are
not blocked, to prevent overheating.
6. Check the voltage at the power point from where the supply is to be given to the system. It should be as
per the specifications. Earth the system properly. (Refer “How to Make the Telecom Earth”)
Inserting Cards
7. Make sure that the ETERNITY power is off and the power cord is unplugged.
8. Open the enclosure slot covers by pressing down the snap lugs.
10. Unscrew and remove the filler bracket that covers the card-slot opening of the slot you intend to use.
11. Hold the card with the connectors facing you. Do not grab the card from both ends.
12. Slide the card into the slot, along the guide rails provided for each slot at the top and bottom planes.
13. Ensure that the cards are inserted deep enough for all the connector pins on the cards make complete
contact with those of the motherboard on the backplane.
14. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
secure the card with the screw provided.
16. Following the above steps, install each card into the universal slots.
Detailed installing instructions are provided for each card - Power Supply Card, Switch Card, DKP, SLT,
TWT, ISDN BRI, ISDN T1E1PRI, GSM, VoIP, E&M, etc. - later in this section. Refer to them when installing
each card type.
• If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
• Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
18. Connect the cables supplied with the cards and lead the cables through the cable guides provided below
the slots in the enclosure. This will ensure neat and tangle-free cabling.
The color markings make it easy to identify the connector and the ports to which the cable is connected.
19. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
20. Close the enclosure cover, pressing down the snap lug as you push each part of the cover in its place.
Two types of Power Supply Cards are supported by the Matrix ETERNITY ME models: PS UNI Card and PS48V
Card.
• PS UNI Card with 100-240VAC, 47-63Hz Mains as Input AC Voltage Power Supply.
This card is designed on the SMPS scheme. As this card does not have any provision for battery backup,
it is recommended that a UPS be connected to keep the system powered during outages.
This card has four LEDs, a Mains Switch, and a Socket assembly for connecting the mains cord.
• PS48V Card with 48VDC as Input DC Power Supply Voltage. A Float cum Boost Charger (FCBC) is
required to feed 48VDC power to the card. The FCBC works on input AC mains.
The card has four LEDs, an MCB Switch, a power ON/OFF Switch, and a 3-way termination block for
connecting the power cord.
Both, the PS UNI card and the PS48V Card provide DC output voltages as: +3.5V, +5.0V, -27V and -85V.
These are indicated by LEDs.
• The ETERNITY provides Redundancy option for the Power Supply card only in the ETERNITY ME10S
variant and for the PS48V card only.
• The ETERNITY ME10S model supports two PS48V power supply cards. Whenever there is a fault in
one, the other takes over the control, providing uninterrupted communication.
• The maximum number of ports supported by the GSM, SLT and DKP Cards may vary according to the
type of Power Supply used. Refer the following table for maximum ports supported with Universal
Power (PSUNI) and DC Power Supply.
Analog SLT ports supported for Short Loop with Loop Current programmed
Number of ports supported in talk mode (OFF-Hook short loop) 250 200 172 150 128
according to loop current programmed
The Power Supply Card is delivered factory fitted, when you buy the system. However, if you want to remove the
card for the purpose of maintenance or replace it with a new one, please follow the instructions below:
1. Unpack the Power Supply Card and verify the package contents.
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
To install a second PS48V card on the ETERNITY ME10S for redundancy, insert the second card on the
next slot. Also refer the topic “Hardware Overview” in Know Your ETERNITY.
5. If installing the PSUNI card, connect the three-pin power cord into the socket of the PS UNI card and plug
in the cord into the mains supply.
You may connect the PSUNI Card to a UPS to keep the system live during power outages.
Select a UPS considering the typical power consumption of ETERNITY presented in the table below:
Power Consumption
Model
(Typical)
6. If installing the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from
the FCBC output into the 3-way termination block on the PS48V card.
Color Signal
Red +48VDC
Black GND
Green Earth
It is recommended that you measure the voltage before connecting the power cable to the power supply
card. Ensure that the earth is connected.
48V Battery
If two PS48V cards are installed for redundancy (possible in ETERNITY ME10S only), each must be
connected to a separate FCBC and each FCBC must be connected to a separate source of power supply.
12. When the batteries are drained, the FCBC goes into the boost mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery volt-
age goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to boost mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the bat-
teries are charged with higher current initially.
The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. If 48V/26Ah
batteries are connected to the FCBC for the ETERNITY ME 10S system then backup time of 2.5 to 3 hrs
can be ensured. The FCBC uses the constant voltage charging method. So, the batteries get charged
faster if less power is consumed by the system when in mains mode.
As the main card in the ETERNITY ME, the Master card manages the entire system. Equipped with a primary and
a secondary controller (also called the Communication Manager), the Master card controls all the slave cards of the
ETERNITY ME: the Switch card, DKP card, SLT card, TWT+SLT card, E&M card, ISDN T1E1PRI card, ISDN BRI
card, VMS card, etc. All the configuration information is stored on this card.
The Master Card occupies a fixed slot in the ETERNITY ME. As the slot designated for the Master Card has a
unique arrangement of connectors, no other card can be inserted into this slot.
The Master card is equipped two Communication Ports, a Digital Input Port, a Digital Output Port, Ethernet Port, a
Printer Port, and three dual color LEDs, on the front panel. Each of these is described briefly below.
Communication Ports
There are two asynchronous, serial, full-duplex RS-232C Communication (COM) Ports, labeled as COM1 and
COM2. The COM Ports have two identical DB-9 connectors.
The COM port allows you to connect a PC to the ETERNITY, so that you can install and operate the following
features:
13
Master Ethernet Port
The Ethernet Port on the fascia of the Master Card is provided to connect ETERNITY to a PC or a LAN to operate
the web-based configuration software Jeeves and the Property Management Software (PMS) for Hotel Application.
A cable with a standard RJ45 connector is provided for the Master Ethernet Port.
Printer Port
The Printer port, labeled PRN, on the Master Card is an industry standard Centronics port with a DB-25 female
connector. You can connect any standard printer. The system sends data in the pure ASCII format. No special
characters or control sequences are sent.
13. The Ethernet port is supported on Master Cards with PCB version V3R0 onwards.
Ensure that the devices connected to the DOP comply with the technical specifications. Refer the topic “Digital
Output Port (DOP)” to know more.
Communication DB-9 female Fascia To connect a PC (to run various software, download
(COM Port) system activity reports).
Communication DB-9 female Fascia To connect a PC (to run various software, download
(COM Port) system activity reports).
Ethernet Port RJ45 Fascia To connect ETERNITY to a PC/LAN (to run the
configuration tool, Jeeves, and Property
Management Software).
Digital Output VDC Max.=60VDC, IDC Fascia To connect automated application devices: hooter,
Port Max.=0.15A, Push-type siren, door lock, fire alarm, bell, water pump, lights.
Connector.
LEDs
The three LEDs located on the Master Card indicate the health of the card during the reset cycle and the health of
the system during its normal functioning. The LED pattern of the Master card is summarized in the table below.
Stage L1 L2 L3
Jumpers
Jumper J9 on the Master Card is used to Reset the SE Password. Refer the table below:
BC Internal Boot
J9 AB Reset SE Password.
BC (default) Normal.
Do not change the position of Jumpers number J8, J10, J12 and J13.
Redundancy for the Master Card is supported only in the ETERNITY ME 10S model. Two Master Cards
can be installed in ETERNITY ME10S. When the active card fails, the standby card takes over control. As
the system restarts during the take over, all existing calls get disconnected. When the standby card
becomes the active card, the system re-boots automatically, restoring communication, all within a few
minutes (2-3 minutes).
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the Master Card into the guide rails of the slot designated for the card.
On the ETERNITY ME10S model, the third and fourth slots from the left are fixed for the Master Card.
(Refer the slot illustrations at the beginning of this topic)
On the ETERNITY ME16S model, the last slot on the right side is designated to the Master Card. (Refer
the slot illustrations at the beginning of this topic)
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
3. If installing a second Master Card on the ETERNITY ME10S for redundancy, insert the second card on the
fourth slot from the left, next to the first Master Card.
4. You can connect the following external devices to the appropriate ports on the ETERNITY Master Card:
Connecting a Printer
5. You can connect any standard printer to the Printer Port (25-pin connector).
Use 0.5mm, non-stranded cables to connect the sensor device to the DIP.
• strip off about half a centimeter of the insulation off the wire ends of the sensor device.
• using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
• insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
• ensure that both wires fit neatly into the opening.
• release pressure on the levers. Both wires will be held in place by spring clamp action.
If you are using ETERNITY ME10S with Redundancy Option, connect the DIP on the Active and the
Standby Master Cards as illustrated below.
A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device can be
interfaced with the DOP via this DC contactor.
The DOP has a two-wire, push-in (spring clamp action) connector to attach the relay device.
Operation Time 5 ms
• strip off about half a centimeter of the insulation off the wire ends of the gadget.
• using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
• insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
• ensure that both wires fit snugly into the openings.
• release pressure on the levers. Both wires will be held in place by spring clamp action.
If you are using ETERNITY ME10S with Redundancy Option, connect the DOP on the Active and the
Standby Master Cards as illustrated below.
The Ethernet Port is located on the Master Card on ETERNITY ME. With the ETERNITY connected to a
LAN, you can:
• access the web-based configuration tool Jeeves from any PC on the LAN.
• set up and run software applications such as PMS and CAS on any PC on the LAN.
• generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
There are two ways to do it, depending on the type of application you are going to use:
9. Connect the Ethernet Port of ETERNITY with the Ethernet Port of the stand-alone PC using the Ethernet
cable supplied with ETERNITY.
You need to connect to the PC via the Ethernet port for the following functions:
10. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
any standard Communication Cable.
When you connect the ETERNITY ME to a standalone/LAN PC, you need to make sure that
• The IP Address of the Master Ethernet Port of the ETERNITY ME and the Ethernet Port of the PC do
not conflict, are not the same.
• The Master Ethernet Port of ETERNITY ME and the Ethernet Port of the PC are in the same Subnet.
If the system is connected to a LAN PC, ask the LAN Administrator to assign an IP Address and a Subnet
Mask to the ETERNITY ME16.
13. Change the IP Address and the Subnet Mask of the Master Ethernet Port by dialing the following
commands from a station of the ETERNITY ME.
• Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
You get programming tone.
To change IP Address
If you have Redundancy option in the Master Card on your ETERNITY ME10S, all configuration settings
must be updated on both the Master Cards by the System Engineer via Jeeves, so that when the standby
card takes over, the system will function with the same configuration settings as the first card. Thus
ensuring smooth take over by the redundant (second) card.
The Switch Card supports the Master Card in performing the functions of speech and data connections, Call
Progress Tone Generation (dial tone, ring back tone, busy tone, etc.), and music generation.
The Switch Card too occupies a fixed slot with a unique arrangement of connectors, so that no other card can be
inserted in this slot.
In the ETERNITY ME16S model, the Switch Card slot is the second last slot on the right. In the ETERNITY ME10S
model, the fifth slot is reserved for the Switch Card. (Refer the slot illustration at the beginning of this topic)
The card has an Analog Input Port, an Analog Output Port, four Digital Key Phone (DKP) ports. All these ports are
connected via a single Amphenol connector (24-Way, 10-Pair) and an MDF cable is supplied with the card.
The two LEDs on the Switch Card to indicate the health of the card during the reset cycle and the status of the DKP
ports during normal functioning of the system.
Music from the external source can be played as Music-on-Hold to internal as well as external callers. Refer the
topics “Background Music (BGM)” and “Music on Hold (MOH)”.
Ensure that the external music source you connect to the AIP complies with its technical specifications.
When you connect an external paging device to the AOP, ensure that it complies with the technical specifications/
requirements of the port.
Analog Input Port Amphenol (24 pin) 0.7Vmrs, Fascia To connect an External Music
Isolated Source
Analog Output Amphenol (24 pin) 0.7Vmrs, Fascia To connect a Public Address
Port Isolated System.
DKP Ports (1-4) Amphenol (24 pin) Fascia To connect digital key phones
• Switch Card Redundancy option is supported only in the ETERNITY ME 10S model. An additional
Switch Card can be installed for redundancy.
• If your ME10S has Switch Card Redundancy and you have connected Digital Key Phones to the DKP
ports on the main card Switch Card, you must have the requisite wiring in place to ensure that the
digital key phones continue to work when the redundant Switch card takes control.
• Refer the following wiring diagram to install the digital key phones with the main Switch Card and the
Redundant card.
• If you are using a new Switch Card (1000 ports) of firmware version V5Rx, use a Master Card with
software version less than or equal to 'V6R10' only.
• If you are using old Switch card (512 ports) of firmware V4Rx, you can use a Master Card with any
software version. (However, in 7th slot, use the cards with configuration of less than 16 ports).
• If you are using a DSP based Switch card it is better to use the firmware version of 'V5R1' to avoid
compatibility issues with software version of Master Card in use. (If you are using firmware version of
Switch card less than 'V4Rx', the cards in the 7th and 8th slot will not work. However, in 7th slot, you
can use the cards with configuration, less than 16 ports).
For example, if you are using Switch card with software version 'V4R2', change it to 'V5R1'.
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the Switch Card into the guide rails of the slot designated for the card. On the ETERNITY ME10S
model, the fifth and the sixth slots from the left are reserved for the Switch Card. On the ETERNITY
ME16S model, the second last slot on the right side is designated to the Switch Card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
3. If installing a second Switch Card on the ETERNITY ME10S for redundancy, insert the second card on the
sixth slot from the left.
4. Use the 24 Way-10 Pair (Amphenol connector) MDF cable supplied with the Switch Card to connect the
following devices to the Switch Card:
Refer the illustration below for the pinout details to help you identify the ports.
The cable pinout details for the DKP Ports on the Switch Card are shown in the above figure.
Terminate the free end of the wires of the DKP Ports into the Main Distribution Frame. Crimping each wire
into the punch down block of the Krone module. Also, refer “The Main Distribution Frame (MDF)”,
“Installing Digital Key Phone Card”.
Connect a good quality external amplifier and matching speakers to the port
Specification Value
• join the stripped end of the amplifier wires with the free end of the wires of the Analog Output Port - the
Blue-Red wire pair - in the Switch Card cable.
Use shielded cable for connecting the amplifier with the speakers.
Specification Value
• strip off about half a centimeter of insulation of the wire-pair of the external music device.
• join the stripped ends of the device wires with the free end of the wires of the Analog Input Port - the Grey-
White wire pair - emerging from the Switch Card cable.
Also refer the topics “Background Music (BGM)”, “Music on Hold (MOH)”, “External Music”.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
a. The current LED state will remain the same until the next command is received
from the application on the DKP Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command is
received it will be turned Green/Red ON again. This process continues.
a. The current LED state will remain the same until the next command is received from
the application on the DKP Port. For example, if the current LED state is Green/Red
ON, on the next command received, the LED will be turned OFF. It will remain OFF
until the next command is received. When the next command is received it will be
turned Green/Red ON again. This process continues.
The Single Line Telephone (SLT) Card provides the interface to connect as extension phones, any standard, two-
wire, analog single line telephone instrument - rotary, pulse-tone, cordless, feature phones with or without Calling
Line Identification.
The SLT Card is available in the following configurations for the variants of ETERNITY ME. SLT interface also is
available in combination with TWT Trunks on a single card.
ETERNITY ME Card Combination card, with 8-ports to connect to 8 Two-wire Analog trunk lines and
TWT8+SLT24 24 Single Line Telephones
ETERNITY ME Card SLT8- Combination card, with 8-ports to connect to 8 Single Line Telephones and 8
MAGNETO8 Magneto Telephones (with Port Swapping)
Choose an SLT Card with the configuration that meets your requirement for SLT ports. Also consider the maximum
SLT Port capacity of the system you are installing. The maximum number of SLT ports supported by the variants of
ETERNITY ME are:
Connectors
The SLT Cards have RJ45 connectors, with each connector having 4 SLT ports. A multi-pair, MDF cable is supplied
for each connector.
LEDs
The SLT cards for ETERNITY ME models have a single, tri-color LED to indicate:
• the status of any one of the station ports during normal functioning of the system.
You may monitor any of the SLT Station ports by assigning the LED to that port17.
17. To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED
Number is the number of the LED on the card, which will monitor the port.
1. Decide the number of SLT extensions required and arrange for as many telephone instruments.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require
you to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of
the phone to dial Flash.
2. Unpack the SLT card and check the package contents. Ensure that the power supply is switched off,
before you begin the installation of the card. Always wear an electrostatic discharge prevention wrist strap/
belt and use a grounding mat.
The SLT Card supports Hot Swap. So, you can insert the SLT Card while the system is switched on.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket! You may require it at a later stage.
4. Insert the SLT Card into the guide rails of the free slot you selected for the card.
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
bracket with the two screws provided.
If you are installing more than one SLT card, you can install the second card in any other free slot. It is not
necessary to install the second/third card in the subsequent slots.
6. Use the cables supplied with the SLT card to connect the SLT wires with the Main Distribution Frame.
For each connector on the SLT Card, there is a separate 4-pair cable with an RJ45 jack on one end and
free at the other end.
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
(Blue) Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2 Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
RJ45-3 Blue - (Blue & White) SLT 09
(Green) Orange - (Orange & White) SLT 10
3 Green - (Green & White) SLT
Brown - (Brown & White) SLT 12
RJ45-4 Blue - (Blue & White) SLT 13
(Brown) Orange - (Orange & White) SLT 14
4 Green - (Green & White) SLT
Brown - (Brown & White) SLT 16
RJ45-5 Blue - (Blue & White) SLT 17
(Blue) Orange - (Orange & White) SLT 18
5 Green - (Green & White) SLT
Brown - (Brown & White) SLT 20
RJ45-6 Blue - (Blue & White) SLT 21
(Orange) Orange - (Orange & White) SLT 22
6 Green - (Green & White) SLT
Brown - (Brown & White) SLT 24
RJ45-7 Blue - (Blue & White) SLT 25
(Green) Orange - (Orange & White) SLT 26
7 Green - (Green & White) SLT
Brown - (Brown & White) SLT 28
RJ45-8 Blue - (Blue & White) SLT 29
(Brown) Orange - (Orange & White) SLT 30
8 Green - (Green & White) SLT
Brown - (Brown & White) SLT 32
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
(Blue) Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2 Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
RJ45-3 Blue - (Blue & White) SLT 09
(Green) Orange - (Orange & White) SLT 10
3 Green - (Green & White) SLT
Brown - (Brown & White) SLT 12
RJ45-4 Blue - (Blue & White) SLT 13
(Brown) Orange - (Orange & White) SLT 14
4 Green - (Green & White) SLT
Brown - (Brown & White) SLT 16
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
1 (Blue) Orange - (Orange & White) SLT 02
Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2
Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
(Blue) Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2 Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
RJ45-3 Blue - (Blue & White) SLT 09
(Green) Orange - (Orange & White) SLT 10
3 Green - (Green & White) SLT
Brown - (Brown & White) SLT 12
RJ45-4 Blue - (Blue & White) SLT 13
(Brown) Orange - (Orange & White) SLT 14
4 Green - (Green & White) SLT
Brown - (Brown & White) SLT 16
RJ45-5 Blue - (Blue & White) SLT 17
(Blue) Orange - (Orange & White) SLT 18
5 Green - (Green & White) SLT
Brown - (Brown & White) SLT 20
RJ45-6 Blue - (Blue & White) SLT 21
(Orange) Orange - (Orange & White) SLT 22
6 Green - (Green & White) SLT
Brown - (Brown & White) SLT 24
RJ45-7 Blue - (Blue & White) TWT 01
(Green) Orange - (Orange & White) TWT 02
7 Green - (Green & White) TWT
Brown - (Brown & White) TWT 04
RJ45-8 Blue - (Blue & White) TWT 05
(Brown) Orange - (Orange & White) TWT 06
8 Green - (Green & White) TWT
Brown - (Brown & White) TWT 08
7. Plug in the RJ45 end of the MDF cables supplied with the card into the respective connectors with the help
of the color markings on the cables as illustrated above for each SLT Card type.
8. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the “The Main Distribution Frame (MDF)”.
9. Repeat the same steps to install another SLT card. Connect the SLT instruments, as described in the next
step. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the
LED Pattern of the SLT Card.
Auto Upgradationa
Initialization
Errors
a. The current LED state will remain the same until the next command is re-
ceived from the application on the SLT Port. For example, if the current LED
state is Green/Red ON, on the next command received, the LED will be
turned OFF. It will remain OFF until the next command is received. When the
next command is received it will be turned Green/Red ON again. This pro-
cess continues.
• For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.
• When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the first port on the
connector will be assigned to the SLT.
• If you have installed SLT8-Magneto8 combination card, it is possible to swap ports and test the
functioning of ports. Refer the topic “Configuring Magneto Interface” for instructions.
For the Building Intercom application, ETERNITY supports the Intercom Line Card (ILC).18
You can connect any standard, two-wire, analog single line telephone instrument - rotary, pulse-tone, cordless,
feature phones with or without Calling Line Identification to the Intercom Line card.
The Intercom Line Card is available in the following configurations for the variants of ETERNITY ME.
Choose an ILC Card with the configuration that meets your requirement for intercom ports. Also, consider the
maximum Port capacity of the system you are installing. The maximum number of intercom ports supported by the
variants of ETERNITY ME are:
Connectors
The ILC Cards have RJ45 connectors with four ports on each connector. A multi-pair, MDF cable is supplied for
each connector.
LEDs
The ILC cards for ETERNITY ME have a single, tri-color LED to indicate the health of the card during the Reset
Cycle.
1. Decide the number of intercom extensions required and arrange for as many telephone instruments.
2. Ensure that the extension wiring is completed according to your requirements. The extension cables from
the wall jack are terminated in the Main Distribution Frame and the telephones are connected to the wall
jacks.
3. Always wear an electrostatic discharge prevention wrist strap/belt and use a grounding mat to prevent
damage to the components of the card.
18. Check Availability. This card is supported by Firmware Version 10.06 and later only.
5. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Keep the filler
bracket for future use.
6. Insert the ILC card into the guide rails of the free slot you selected for the card.
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
7. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
bracket with the two screws provided.
9. Now, use the cables supplied with the ILC card to connect the card to the Main Distribution Frame to which
the intercom phones are connected.
For each connector on the card, there is a separate 4-pair cable with an RJ45 jack on one end and free at
the other end. Refer the illustrations of the pinout of the intercom cards to connect the wires.
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
(Blue) Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2 Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
RJ45-3 Blue - (Blue & White) SLT 09
(Green) Orange - (Orange & White) SLT 10
3 Green - (Green & White) SLT
Brown - (Brown & White) SLT 12
RJ45-4 Blue - (Blue & White) SLT 13
(Brown) Orange - (Orange & White) SLT 14
4 Green - (Green & White) SLT
Brown - (Brown & White) SLT 16
RJ45-5 Blue - (Blue & White) SLT 17
(Blue) Orange - (Orange & White) SLT 18
5 Green - (Green & White) SLT
Brown - (Brown & White) SLT 20
RJ45-6 Blue - (Blue & White) SLT 21
(Orange) Orange - (Orange & White) SLT 22
6 Green - (Green & White) SLT
Brown - (Brown & White) SLT 24
RJ45-7 Blue - (Blue & White) SLT 25
(Green) Orange - (Orange & White) SLT 26
7 Green - (Green & White) SLT
Brown - (Brown & White) SLT 28
RJ45-8 Blue - (Blue & White) SLT 29
(Brown) Orange - (Orange & White) SLT 30
8 Green - (Green & White) SLT
Brown - (Brown & White) SLT 32
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
(Blue) Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2 Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
RJ45-3 Blue - (Blue & White) SLT 09
(Green) Orange - (Orange & White) SLT 10
3 Green - (Green & White) SLT
Brown - (Brown & White) SLT 12
RJ45-4 Blue - (Blue & White) SLT 13
(Brown) Orange - (Orange & White) SLT 14
4 Green - (Green & White) SLT
Brown - (Brown & White) SLT 16
10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle and the
LED indication of the card.
Auto Upgradationa
Initialization
Errors
The Digital Key Phone (DKP) Card provides the interface to connect the proprietary digital key phones, EON, the
PC-based phone EONSOFT and the Direct Station Selection (DSS) Consoles with the ETERNITY.
The DKP Card is available in the following configurations for the models of ETERNITY ME.
Select a DKP Card with the configuration that meets your requirement for DKP Ports. Also consider the maximum
DKP Port capacity of the system you are installing.
Both ETERNITY ME 10S and ME16S support a maximum of 128 DKP Ports.
• If you have used up the four in-built DKP ports on the Switch Card, you can connect a maximum of 124
DKPs.
Connectors
The DKP Cards have RJ45 connectors, with each connector having 4 DKP ports. A multi-pair MDF cable is
supplied for each connector on the card.
LEDs
The DKP card has two dual color LEDs:
• LED1 indicates the health of the card during the Reset Cycle.
• LED2 monitors the status of any one of the station ports during normal functioning of the system.
LED 2 can be assigned to any DKP port to monitor the status of that port19.
1. Decide the number of DKP extensions and DSS Consoles required and arrange for as many EON,
EONSOFT and DSS Consoles.
19. You can do this from the SE mode, by dialing the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the uni-
versal slot in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its function-
ing. LED Number is the number of the LED on the card, which will monitor the port.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
4. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
with the two screws provided.
If you are installing more than one DKP card, it is not necessary to install the next card in the subsequent
slots.
6. Using the MDF Cables supplied with the DKP card connect the DKP wire pairs to the Main Distribution
Frame.
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) DKP 01
(Blue) Orange - (Orange & White) DKP 02
1 Green - (Green & White) DKP
Brown - (Brown & White) DKP 04
RJ45-2 Blue - (Blue & White) DKP 05
(Orange) Orange - (Orange & White) DKP 06
2 Green - (Green & White) DKP
Brown - (Brown & White) DKP 08
RJ45-3 Blue - (Blue & White) DKP 09
(Green) Orange - (Orange & White) DKP 10
3 Green - (Green & White) DKP
Brown - (Brown & White) DKP 12
RJ45-4 Blue - (Blue & White) DKP 13
(Brown) Orange - (Orange & White) DKP 14
4 Green - (Green & White) DKP
Brown - (Brown & White) DKP 16
RJ45-5 Blue - (Blue & White) DKP 17
(Blue) Orange - (Orange & White) DKP 18
5 Green - (Green & White) DKP
Brown - (Brown & White) DKP 20
RJ45-6 Blue - (Blue & White) DKP 21
(Orange) Orange - (Orange & White) DKP 22
6 Green - (Green & White) DKP
Brown - (Brown & White) DKP 24
RJ45-7 Blue - (Blue & White) DKP 25
(Green) Orange - (Orange & White) DKP 26
7 Green - (Green & White) DKP
Brown - (Brown & White) DKP 28
RJ45-8 Blue - (Blue & White) DKP 29
(Brown) Orange - (Orange & White) DKP 30
8 Green - (Green & White) DKP
Brown - (Brown & White) DKP 32
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) DKP 01
(Blue) Orange - (Orange & White) DKP 02
1 Green - (Green & White) DKP
Brown - (Brown & White) DKP 04
RJ45-2 Blue - (Blue & White) DKP 05
(Orange) Orange - (Orange & White) DKP 06
2 Green - (Green & White) DKP
Brown - (Brown & White) DKP 08
RJ45-3 Blue - (Blue & White) DKP 09
(Green) Orange - (Orange & White) DKP 10
3 Green - (Green & White) DKP
Brown - (Brown & White) DKP 12
RJ45-4 Blue - (Blue & White) DKP 13
(Brown) Orange - (Orange & White) DKP 14
4 Green - (Green & White) DKP
Brown - (Brown & White) DKP 16
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) DKP 01
1 (Blue) Orange - (Orange & White) DKP 02
Green - (Green & White) DKP
Brown - (Brown & White) DKP 04
RJ45-2 Blue - (Blue & White) DKP 05
(Orange) Orange - (Orange & White) 06
2 DKP
Green - (Green & White) DKP
Brown - (Brown & White) DKP 08
7. Plug in the RJ45 end of the DKP cables into the respective connectors guided by the color markings on the
cables as illustrated above for each DKP Card Type.
8. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY ME DKP Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the extension line to be connected to this port must be terminated on the
top of the Krone connector. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
9. Connect the Digital Key Phone to the DKP Ports as described in the next step. If you have completed all
installation tasks, power on the system and observe the Reset Cycle and the LED Pattern of the DKP
Card.
Auto Upgradation
Initialization
Errors
Selected DKP's data are transmitted to Master Card RED Togglea on each event
Selected DKP's data are received from Master Card RED Toggleb on each request from Master
a. The current LED state will remain the same until the next event is received from the application on the DKP Port.
For example, if the current LED state is Green/Red ON, on the next event, the LED will be turned OFF. It will
remain OFF until the next event occurs. When the next event is received it will be turned Green/Red ON again.
This process continues.
b. Same as above note.
Installing EON
Matrix offers EON, the proprietary digital key phone. EON is available in the following models:
• EON42
• EON48
• Now, insert the snap fits of the foot stand into the Wall Mount bracket slots on the bottom of the phone in
the " wall up" direction.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON42. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
• Connect the handset of the EON42 to the phone body using the spring cord.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
EON48
• To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
• Now, mount the phone with the screws fitting into the keyhole slots.
• When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
• Connect the handset of the EON48 to the phone body using the spring cord.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
You can also plug in a headset with RJ11 connector into the Headset port at the bottom of the phone.
3. Plug one end of the RJ45 cable supplied with the phone into the RJ45 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
4. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait'… appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PL EASE WAI T .. .
202 Reception
M on 2 4 A U G 1 2 : 0 0
6. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, “Digital Key Phone-
Operation”.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
2. Place the DSS Console next to the DKP, EON, to which it is to be attached.
You can install two DSS consoles to a DKP. Refer “Direct Station Selection Console” for possible
combinations for installing the various models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated into the bottom of
the Krone Connector (of 'Station Lines' on the MDF).
5. The wire-pairs of the DSS Consoles should be terminated into the top of the Krone Connector (of 'Station
Lines' on the MDF).
6. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
7. ETERNITY automatically assigns the first DSS Console discovered on the system to the first DKP, the
second DSS to the second DKP.
8. Only when two DSS Consoles are to be assigned to a single DKP, manual assignment of DSS to the DKP
is required. Refer “Configuring DKP Extensions”.
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
the speaker connectors into the dongle.
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
communication cable into the COM port of the computer.
4. Connect a wire-pair of a DKP port of the ETERNITY to the RJ11 port marked 'DKP' on the dongle.
5. Switch ON the computer. The computer must have Windows Operating System installed on it. If you do not
have of the operating systems mentioned above, install any compatible Windows Operating System.
6. Now insert the EONSOFT CD-ROM supplied with this PC-based DKP into the CD drive of your Computer.
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
12. Click 'Options' at the top left of the window. A drop down menu will appear.
14. Select the COM Port to which the communication cable is connected.
This screen will appear only if the DKP port to which the EONSOFT is connected has been programmed for
parameters like Name, Station number, Date and Time.
• If this dialog box does not appear on the screen in response to the click to select the COM Port Option,
test the COM Port for data transfer.
• If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
• Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
• Click the button labeled 'Start Test' in the COM Port Settings dialog box.
• After clicking this button, observe the Test Result section on the dialog box.
The above figure shows that the COM Port/communication cable is working.
• If the 'Error Count' shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
• Remove the communication cable from the COM Port of the PC.
• Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
• Now, if the error count is zero, please check the Communication Cable.
• If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
The Two-Wire Trunk (TWT) Card provides the interface to connect the ETERNITY with the POTS Network. The
TWT Card supports the different standards and features of POTS Networks across the world.
The TWT Card is available in the following configurations for the variants of ETERNITY ME. TWT interface is also
available in combination with SLT ports on a single card.
ETERNITY ME Card TWT16 16-port card to connect 16 Two-wire Trunk lines from the CO network
ETERNITY ME Card TWT8 8-port card to connect 8 Two-wire Trunk lines from the CO network
ETERNITY ME Card TWT8+SLT24 Combination card, with 8 TWT ports to connect 8 TWT analog trunk lines
and 24 SLT ports to connect 24 Single Line Telephones
ETERNITY ME Card Combination card, with 8 TWT ports to connect 8 TWT analog trunk lines
TWT8+MAGNETO8 and 8 Magneto ports to connect 8 Magento Telephones (with port
swapping)
Choose a TWT Card with the configuration that meets your requirement for TWT trunk ports, keeping in mind the
maximum TWT Trunk Port capacity of the system you are installing.
ETERNITY ME 10S and ME16S both support a maximum of 128 TWT Ports.
Connectors
The TWT Card has RJ45 connectors, with 4 TWT ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
LED
The TWT Cards have a single tri-color LED to indicate:
• the status of a selected Trunk port during normal functioning of the system.
You can assign the LED to any TWT port on the card which you want to monitor20.
20. To assign the LED to a selected port for monitoring its functioning, you must enter SE mode and dial the SE Command 7902-Slot-
LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor
the port.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Preserve the filler bracket for future use!
4. Insert the TWT Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
5. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one TWT Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
6. Use the cables supplied for each connector on the TWT card to connect the Trunk Lines with the Main
Distribution Frame.
The cable for each connector is identified by a distinct color marked at the Boot edge and the Insulation
edge of the cable. This is to help you identify the connector and the ports to which the cable is connected.
You may refer the illustrations below to identify which cable to plug into each connector with the help of the
color markings on the cable.
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) TWT 01
(Blue) Orange - (Orange & White) TWT 02
1 Green - (Green & White) TWT
Brown - (Brown & White) TWT 04
RJ45-2 Blue - (Blue & White) TWT 05
(Orange) Orange - (Orange & White) TWT 06
2 Green - (Green & White) TWT
Brown - (Brown & White) TWT 08
RJ45-3 Blue - (Blue & White) TWT 09
(Green) Orange - (Orange & White) TWT 10
3 Green - (Green & White) TWT
Brown - (Brown & White) TWT 12
RJ45-4 Blue - (Blue & White) TWT 13
(Brown) Orange - (Orange & White) TWT 14
4 Green - (Green & White) TWT
Brown - (Brown & White) TWT 16
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) TWT 01
1 (Blue) Orange - (Orange & White) TWT 02
Green - (Green & White) TWT
Brown - (Brown & White) TWT 04
RJ45-2 Blue - (Blue & White) TWT 05
(Orange) Orange - (Orange & White) TWT 06
2
Green - (Green & White) TWT
Brown - (Brown & White) TWT 08
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
(Blue) Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04
RJ45-2 Blue - (Blue & White) SLT 05
(Orange) Orange - (Orange & White) SLT 06
2 Green - (Green & White) SLT
Brown - (Brown & White) SLT 08
RJ45-3 Blue - (Blue & White) SLT 09
(Green) Orange - (Orange & White) SLT 10
3 Green - (Green & White) SLT
Brown - (Brown & White) SLT 12
RJ45-4 Blue - (Blue & White) SLT 13
(Brown) Orange - (Orange & White) SLT 14
4 Green - (Green & White) SLT
Brown - (Brown & White) SLT 16
RJ45-5 Blue - (Blue & White) SLT 17
(Blue) Orange - (Orange & White) SLT 18
5 Green - (Green & White) SLT
Brown - (Brown & White) SLT 20
RJ45-6 Blue - (Blue & White) SLT 21
(Orange) Orange - (Orange & White) SLT 22
6 Green - (Green & White) SLT
Brown - (Brown & White) SLT 24
RJ45-7 Blue - (Blue & White) TWT 01
(Green) Orange - (Orange & White) TWT 02
7 Green - (Green & White) TWT
Brown - (Brown & White) TWT 04
RJ45-8 Blue - (Blue & White) TWT 05
(Brown) Orange - (Orange & White) TWT 06
8 Green - (Green & White) TWT
Brown - (Brown & White) TWT 08
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors referring to the color
markings on the cables as illustrated above for each TWT Card type.
8. Terminate the free end of the TWT Card cable into the punch down blocks of the Krone modules
designated for 'Trunk Lines' on “The Main Distribution Frame (MDF)”.
Trunk cables from the ETERNITY are to be connected with the Trunk Lines from the PSTN/CO terminated
on the MDF.
Refer the topics “The Main Distribution Frame (MDF)” and “Terminating Trunk and Station Cables on the
MDF”.
10. If you have completed all other installation tasks, power ON the system,
Auto Upgradationa
Initialization
Errors
Selected TWT's data are received RED Toggleb on each request from Master
from Master Card
a. The current LED state will remain the same until the next event is received from the application on
the TWT Port. For example, if the current LED state is Green/Red ON, on the next event, the LED
will be turned OFF. It will remain OFF until the next event occurs. When the next event is received
it will be turned Green/Red ON again. This process continues.
b. Same as above note.
If you have installed TWT8-Magneto8 combination card, it is possible to swap ports and test the
functioning of ports. Refer the topic “Configuring Magneto Interface” for instructions.
The BRI card provides the interface to connect ETERNITY with ISDN BRI Lines. The BRI lines may be from a
public ISDN exchange or a private ISDN exchange.
The BRI Card is available in the following configurations for the variants of ETERNITY ME.
ETERNITY ME Card BRI8 8-Port card to connect 8 ISDN BRI Lines or ISDN Compatible Devices
ETERNITY ME Card BRI4 4-Port card to connect 4 ISDN BRI Lines or ISDN Compatible Devices
Connectors
The BRI cards have RJ45 Connectors. The ETERNITY ME BRI8 card has 8 RJ45 connectors for 8 BRI ports.
The ETERNITY ME BRI4 card has 4 RJ45 connectors for 4 BRI ports. A separate cable is supplied for each
connector.
LEDs
The ETERNITY ME BRI8 has 8 LEDs and BRI4 has 4 LEDs.
ISDN
Network NT 1 BRI Port
ETERNITY
Power
U-Interface S/T
(2-wire) Interface
Customer Premises
Where,
• U Interface = between the NT1 equipment and the ISDN central office.
• S/T Interface = between the ISDN user equipment, that is, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the
ETERNITY.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
• the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
• the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
Point-to-Point Configuration
BRI Line
NT BRI Port
ISDN (TE Mode)
Network
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be upto 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
ISDN NT
Network BRI Bus Bar
Terminal
BRI Port Resistance 100
(TE Mode)
Terminal
Resistance 100
ETERNITY ISDN Phone ISDN Phone ISDN Phone
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
• A maximum of 8 TEs or ISDN devices can be connected to a single NT on a bus up to 200 meters from the
NT.
• 100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
• Using this configuration, any subscriber from ETERNITY can access a BRI line and can make outgoing
calls. At the same time, another subscriber from ETERNITY or any ISDN phone shown in the figure can
make outgoing call from the same BRI. In the same way, incoming calls are possible on the same BRI.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d < 1 Km
d1 < 30 meters
ISDN NT
Network BRI Bus Bar
Terminal
Terminal BRI Port Resistance 100
(TE Mode)
Resistance 100
• You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
• However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
• Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
1. Take all the necessary precautions prescribed for handling the cards and electronic equipment: turn off
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
Jumper Position
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
Jumper Position
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
Jumper Position
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
By default, Orientation Type is TE. So, you may skip to the next step.
ISDN
BRI Line NT
Network
• Last TE equipment.
• Last point of the bus bar where the last TE equipment is connected.
• If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
• Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Jumper Position
Function
J3 J4
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position).
1 RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1 1 RJ45 Connector
ports on BRI Bus
Bar to which the
3 3
ISDN TE
4 4 Equipment is
connected
5 5
6 6
8 8
• The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
• Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
• Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
• Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
To do this, you must change the position of the Jumpers J1 and J2 on the BRI modules (daughterboard),
on the BRI Card.
Jumper Position
Function
J1 J2
• To feed the power on Tx and Rx wires, set the jumpers J1 and J2 of BRI module in AB position.
• To feed the power on separate pairs of wires, set the jumpers J1 and J2 of BRI module in BC position.
• To power the ISDN terminal from external power source, keep the jumpers J1 and J2 open.
• The maximum power that can be fed to a single BRI port is 50mA.
• From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
• The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
9. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
bracket in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
4 Tx
5 Rx
3 Rx1
4 Tx1
5 Tx2
6 Rx2
1 -- Orange-White
2 -- Orange
3 TX_A Green-White
4 RX_A Blue
5 RX_B Blue-White
6 TX_B Green
7 VOUT- Brown-White
8 VOUT+ Brown
1 -- Orange-White
2 -- Orange
3 RX_A Green-White
4 TX_A Blue
5 TX_B Blue-White
6 RX_B Green
7 VOUT- Brown-White
8 VOUT+ Brown
11. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle, and the LED pattern of the BRI Card.
• The BRI8 Card has 8 LEDs: L1, L2, L3, L4, L5, L6, L7, L8.
• The BRI4 Card has 4 LEDs: L1, L2, L3, L4.
The LEDs show the Status of the Ports as summarized in the table below:
The ETERNITY T1E1PRI Card provides the interface to connect ETERNITY to ISDN Network.
When connected to T1 carrier lines, the Card supports the following signaling types:
• PRI
• Robbed Bit Signaling
• Q-Signaling (QSIG)
• E&M
When connected to E1 carrier lines, the card supports the following signaling types:
• PRI
• Channel Associated Signaling (CAS)
• Q-Signaling (QSIG)
• E&M
The T1E1PRI Card is available in the following configurations for ETERNITY ME:
ETERNITY ME Card T1E1PRI 2-Port card with QSIG support to connect 2 ISDN T1/E1 PRI Lines or ISDN
Dual Compatible Devices
ETERNITY ME Card T1E1PRI 1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Single Compatible Device
Connectors
The T1E1PRI card has an RJ45 Connector for each port. The ETERNITY ME T1E1PRI Dual card has 2 RJ45
Connectors for the two ports, while the ETERNITY ME T1E1PRI Single card has a single RJ45 Connector.
A cable with RJ45 plugs on both ends is supplied for each connector.
LEDs
The ETERNITY ME T1E1PRI Dual Card has four LEDs: L1, L2, L3 and L4.
The ETERNITY ME T1E1PRI Single Card has two LEDs L1 and L2.
1. Before installing the card, take the necessary precautions prescribed for handling the cards. Always wear
an electrostatic-discharge preventive wrist strap and use a grounding mat. Make sure the power supply is
turned off.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Preserve the filler bracket for future use!
5. By default, termination resistance of PRI port is set as 120, which is for E1 connectivity.
• To use the PRI Port for T1 connectivity, termination resistance must be changed to 100.
• Use DIP Switch SW5 to change the Termination Resistance of PRI Port 1. Set the Pins of SW5 as
shown below:
• If using the ETERNITY ME T1E1PRI Dual Card, use DIP Switch SW2 to change the Termination
Resistance of PRI Port 2. Set the Pins of SW2 as shown below:
6. Insert the T1E1PRI Card into the guide rails of the free slot you selected for the card. Make sure that the
connectors on the card make perfect contact with those of the slot on the backplane motherboard.
7. Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
bracket in place with the two screws provided.
If you are installing more than one T1E1PRI Card, it is not necessary to insert the other cards in
subsequent slots. Any card can be inserted in any of the Universal Slots.
Customer Premises
ETERNITY
ISDN G.703
Modem
Network 4-wire HDSL DTE
4-wire
PRI Port
(RJ-45 Connector) (RJ-45 Connector)
G.703
Modem
Power
• Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line, that is,
one at the Local Exchange and other at the Customer's Premises.
• At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
• The DTE interface of the modem is be connected to the PRI port (RJ-45 connector on the Matrix
ETERNITY ME T1E1PRI Dual/Single Card).
9. Refer the following pin details for connecting the Network Termination Unit with the ETERNITY.
Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)
5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
10. If you have completed all other installation tasks. Power the system. After the Reset Cycle is completed,
observe the LED patterns of the T1E1PRI Card.
The ETERNITY ME T1E1PRI Single Card has two LEDs: L1 and L2.
Given below are the LED Patterns defined for indicating port states in the signaling types supported by the
ETERNITY ME.
LED1/LED3 Pattern:
LED2/LED4 Pattern:
LED1/LED3 Pattern:
LED2/LED4 Pattern:
LED1/LED3 Pattern:
LED2/LED4 Pattern:
Near end loop back wait before activate RED 100ms ON-100 ms OFF
Near end loop back wait before deactivate RED 500ms ON-500 ms OFF
Far end loop back wait after activate GREEN 100ms ON-100 ms OFF
Far end loop back wait after deactivate GREEN 500ms ON-500 ms OFF
LED2/LED4 Pattern:
The Mobile Card interfaces the ETERNITY with GSM/3G networks. It routes calls made and received over mobile
networks, like a mobile handset.
The card does not support GPRS features, Fax and Data services, network supported services, except
CLIR and USSD.
The Mobile card is available in the following configurations for ETERNITY ME.
ETERNITY ME Card GSM8 8-port card to connect to 8 GSM networks (8 SIM Cards can be installed)
ETERNITY ME Card GSM4 4-port card to connect to 4 GSM networks (4 SIM Cards can be installed)
ETERNITY ME Card GSM8 8-port card to connect to 8 GSM networks with 3G support (8 SIM Cards can be
3G installed)
ETERNITY ME Card GSM4 4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards can be
3G installed)
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
Antenna
There is a single rooftop (RT) antenna for four GSM ports. A splitter connects all the four ports on the card into a
single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another position
(in case of weak signal).
LEDs
There is a tri-color LED for each mobile port on the card to indicate the functioning of the card and the status of the
ports.
• If using a GSM/3G card, get the get the SIM Card from the GSM/3G service provider of your choice
ready. Use SIM PIN protection, if required.
2. Make sure that the ETERNITY is installed at a location where sufficient network coverage is available. The
power supply should be turned off, and you must be wearing an electrostatic discharge preventive wrist
strap and a have a grounding mat, before you begin handling the card.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same GSM service provider or of different service providers.
6. Insert the Mobile card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
10. Wait for the system to register with the Mobile network. By default, the Mobile ports are set to select and
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.
• At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
• Each time the Mobile Port sends a request, such as a Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
At Power On: All LEDs will blink 1 second ON and 1 second OFF in the color sequence: Red-Green-Orange until
the Reset cycle is complete.
In the Stand-by state: All LEDs will glow Orange for a second and turn Green for a second, repeatedly.
During normal functioning: The LEDs will various events on the Mobile port in the color and cadence described
in the table below:
SIM PIN faulty Orange 200ms ON-200ms OFF-200ms ON-200ms OFF-200ms ON-2000ms
OFF (3 blinks)
The E&M Card of the ETERNITY ME provides the interface for analog trunking to connect various communication
equipment telephone switches, Routers, Leased Lines, etc. using Tie-Lines.
• Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer “PLCC-An Introduction” to know more.
• “Closed User Group (CUG)”, where several PBXs are connected with each other through E&M tie lines21.
• PBX expansion, where two PBXs are connected with each other with E&M tie lines.
An E&M Port can be programmed to behave as a Trunk Interface, a Subscriber (Station) Interface or both, as a Tie
Line with the dual personality of a Trunk and a Subscriber.
The ETERNITY E&M Card is available in the following configuration for the ETERNITY ME:
Connectors
The E&M8 card is supplied with RJ45 or Amphenol connectors. On the ETERNITY ME E&M8 card with Amphenol
connectors, the first 4 E&M ports (E&M1 to E&M4) are located on the lower connector, and the remaining four E&M
ports (E&M5 to E&M8) are located on the upper connector.
The ETERNITY ME E&M4 card has a single Amphenol connector with 4 ports.
21. The PBXs in a “Closed User Group (CUG)” can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
22. This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
The maximum number of E&M ports supported by each model of ETERNITY ME are:
• ETERNITY ME10S: 80
• ETERNITY ME16S: 128
OR
• a Trunk - works like a trunk interface when any of the stations of the PBX makes an outgoing call through
it.
OR
• a Tie Line - takes on a dual personality: functioning as both a station and a trunk. The E&M port works like
a station interface for incoming calls. It works like a trunk interface when any station makes an outgoing
call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
“PLCC-An Introduction” to know more.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
4. Select the speech interface - 2-wire speech or 4-wire speech - as required, by changing the jumper
settings. Refer the table below.
• By default all the E&M Ports are set to support 2-wire Speech Interface.
• To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
• To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
5. Now, select a free slot for the E&M card. Unscrew and remove the filler bracket by pushing up the levers
on the bracket. Preserve the filler bracket for future use.
6. Insert the E&M Card into guide rails of the empty slot. Make sure the connectors on the card make perfect
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M card into the RJ45/Amphenol connectors on the E&M Card.
8. Connect the other end of the cable into the E&M Ports of the other PBX/Router/Tie Line equipment by
appropriate crossing of the wires.
Refer the following pin-out details for each E&M Card Type and for each E&M Type and Speech Interface
Type.
L1 L5
L2 L6
L3 L7
L4 L8
01 Open Gray 01
02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
1 RJ45-1 05 SPCH A Blue
Pin No. Connection Colour H/w Port Offset 06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 02
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White RJ45-2 2
07 RX SPCH B Orange Pin No. Connection Colour H/w Port Offset
08 E IN Brown-White
09 BGND Brown 01 Open Gray 03
10 CCC Gray-White 02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
3 RJ45-3 05 SPCH A Blue
Pin No. Connection Colour H/w Port Offset 06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 04
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White
RJ45-4 4
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown Pin No. Connection Colour H/w Port Offset
10 CCC Gray-White
01 Open Gray 05
02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
Pin No. Connection Colour H/w Port Offset 5 RJ45-5 05 SPCH A Blue
06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 06
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White RJ45-6 6
07 RX SPCH B Orange Pin No. Connection Colour H/w Port Offset
08 E IN Brown-White
09 BGND Brown 01 Open Gray 07
10 CCC Gray-White 02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
Pin No. Connection Colour H/w Port Offset
7 RJ45-7 05 SPCH A Blue
06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 08
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White RJ45-8 8
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown
10 CCC Gray-White
L1 L3
L2 L4
01 Open Gray 01
02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
1 RJ45-1 05 SPCH A Blue
Pin No. Connection Colour H/w Port Offset 06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 02
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White RJ45-2 2
07 RX SPCH B Orange Pin No. Connection Colour H/w Port Offset
08 E IN Brown-White
09 BGND Brown 01 Open Gray 03
10 CCC Gray-White 02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
3 RJ45-3 05 SPCH A Blue
Pin No. Connection Colour H/w Port Offset 06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 04
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White
RJ45-4 4
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown
10 CCC Gray-White
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the PBX
when speech is established. The E&M Card supports this facility. The ETERNITY sends CCS signal to the
PLCC panel.
• When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
• When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
9. If you have finished all installation tasks, power ON the system, observe the Reset Cycle and the LED
Pattern of the E&M Card.
Initialization
Errors
The Magneto Card is used for connecting the ETERNITY to Magneto Telephones23, which are widely used by the
defense establishments as field phones in front lines, and by other establishments such as railroad companies
(signaling emergencies, crossings, etc.), electric utilities, pipeline companies, who need to have their networks at
places that are too remote to be serviced by public telephone networks.
The Magneto Card of ETERNITY lands calls from magneto field telephones on the Stations of the ETERNITY and
places calls from the Stations of the ETERNITY on magneto telephones.
The Magneto card is available in the following configuration for the ETERNITY ME.
The maximum number of Magneto Ports supported by each variant of ETERNITY ME are:
Connectors
The Magneto Card has RJ45 connectors, with 4 ports on each connector. A multi-pair cable is provided for each
connector.
LED
The ETERNITY Magneto8 has 8 LEDs for each magneto port supported by the card.
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports during the normal
functioning of the system.
You may install an MDF to connect the Magneto Ports with the Field Telephone wires.
OR
You may connect the wires from the Magneto Field Telephones directly to the Magneto Port.
You are advised to use a separate set of Krone Modules for connecting the Magneto phones to the
Magneto ports of ETERNITY.
23. A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator, usually
a magneto. The hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is
turned to produce energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the
mechanical motion via the crank to produce sufficient energy to ring other phones or to signal the CO.
4. Select any universal slot to insert the card. Unscrew the filler bracket and remove it by pushing up the
levers on the bracket.
5. Insert the Magneto card into the guide rails of the free slot. The card's connectors must make perfect
contact with the connectors on the backplane motherboard. Press down the levers of the mounting bracket
to secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
6. Now, plug in the cables supplied with the Magneto Card into the connectors on the card. Terminate the
free ends of the cables into the MDF, if applicable.
Refer to the following block diagram for terminating the cables from the Magneto Card and the wires from
the Magneto Field Telephones.
9. If you do not have any other Card to insert and have completed the installation procedures, plug in the
power cord, switch ON power supply from the mains, switch on the Power supply of the ETERNITY and
observe the Reset Cycle and the LED Pattern of the Magneto Card.
Initialization
Errors
Invalid Card Configuration Jumper ORANGE All LEDs Flash (250ms ON-250ms OFF) twice,
OFF 3 sec.
Controller Xram Failure ORANGE LEDs Flash (250ms ON-250ms OFF) four times,
OFF 3 sec.
Controller Eprom Failure ORANGE LEDs Flash (250ms ON-250ms OFF) 5 times,
OFF 3 sec.
Invalid Slot detectionb ORANGE LEDs Flash (250ms ON-250ms OFF) 6 times,
OFF 3 sec.
The SLT/TWT-Magneto Card is used for connecting Single Line Telephones (SLT), Two-wire Trunks (TWT) and
Magneto Telephones.
You can connect SLT and Magneto Telephones or TWT trunk lines and Magneto Telephones on the same card.
The SLT/TWT-Magneto Card supports Port Swapping, using which you can interchange the SLT/TWT and
Magneto ports. The card also supports BITE (Built-in Test) which allows testing of the functioning of any desired
port on the card.
ETERNITY ME Card SLT8-Magneto8 8-port card to connect 8 SLT and 8 Magneto Phones
ETERNITY ME Card TWT8-Magneto8 8-port card to connect 8 TWT lines and 8 Magneto Phones
Connectors
The SLT/TWT-Magneto Card has RJ45 connectors, with an SLT/TWT and a Magneto port on each connector. A
separate cable is provided for each connector.
LED
The ETERNITY SLT/TWT-Magneto Card has 8 dual color LEDs.
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports after the Reset Cycle.
The LEDs are programmable. You can assign an LED to any port to monitor its functioning24. By default, all 8 LEDs
show the status of the Magneto Ports.
1. Have the necessary wiring for the Magneto Phones, the SLTs and TWT Trunks in place. Prepare for the
card installation by switching off power supply and wearing an electrostatic discharge preventive wrist
strap and use a grounding mat.
24. To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is number of the port on the card to which the LED is to be assigned to monitor its function-
ing. LED Number is the number of the LED on the card, which will monitor the port.
4. Insert the card into the guide rails of the free slot. Make sure the card's connectors make perfect contact
with the connectors on the backplane motherboard. Press down the levers of the mounting bracket to
secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
5. Now, plug in the cables supplied with the SLT/TWT-Magneto Card into the connectors on the card.
Terminate the free ends of the cables into the MDF.
When connecting Magneto telephones, you may either terminate the Magneto wire pairs into the MDF or
you may directly connect the wires from the Magneto Telephones into the Magneto Ports.
Refer to the following pinout details of the card connectors for terminating the cables from the card into the
MDF and the wires from the Magneto Telephones, SLT and TWT trunks.
For detailed instructions, refer the topics “The Main Distribution Frame (MDF)” and “Terminating Trunk and
Station Cables on the MDF”.
Also refer the topics “The Single Line Telephone Card” and “The Two-Wire Trunk Card”.
The SLT/TWT-Magneto card supports Port Swapping. The pin details presented here represent the
'Normal' mode. For pin details in the 'Swapped' mode, refer the topic “Port Swapping of Magneto, SLT and
TWT Ports”.
7. If you have finished installing the desired number of cards, plug in the power cord, switch ON power supply
from the mains, switch on the Power supply of the ETERNITY and observe the Reset Cycle and the LED
Pattern of the SLT/TWT-Magneto Card.
Initialization
Errors
Invalid Slot detectionb ORANGE LEDs Flash (250ms ON-250ms OFF) 6 times, OFF 3 sec.
Port Events:
A Loop Dial (LD) port is a unique port with a dual personality. It is used as a tie-line to connect two PBXs over Two-
wire Trunk lines. The LD Port has a dual personality of a Trunk and a Station.
For outgoing calls, the port assumes the personality of a Two-wire Trunk port. For incoming calls, it behaves like an
SLT port.
The LD Trunk port can be used for expanding the PBX capacity, connecting a second PBX with the main PBX or to
connect two or more remote PBXs with each other, forming a network of PBXs.
The Loop Dial port is available as a combination-card in the following configuration for ETERNITY ME.
The number of LD ports supported by ETERNITY ME depends on the model and the number of Loop Dial
combination cards installed in the system.
Connectors
The SLT8+MAG2+TWT2+LD2+ENM2 Card has 7 RJ45 connectors. A separate cable is provided for each
connector.
LED
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports after the Reset Cycle.
The LEDs are programmable. You can assign an LED to any port to monitor its functioning25.
25. To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is number of the port on the card to which the LED is to be assigned to monitor its function-
ing. LED Number is the number of the LED on the card, which will monitor the port.
For instructions on connecting SLTs, TWT Trunks, Magneto phones and E&M tie lines, refer the respective topics.
Installation Scenario
The LD Trunk port is used for connecting two PBX systems with each other, as illustrated below.
EPBX A EPBX B
LD Port LD Port
CO CO
• The LD port works as a trunk port for extension users, when they make outgoing calls. When the extension
user of PBX A grabs its LD port by dialing a feature access code, it will receive the dial tone of PBX B. The
extension user of PBX A can dial the desired extension number or external number just like any other
extension of PBX B.
• The LD port works as an extension for incoming calls. When the extension user of PBX B grabs the LD
port of PBX B, PBX A feeds dial tone to PBX B over its LD port.
1. Have the necessary wiring in place for LD Ports ready between the desired number of PBXs you want to
interconnect. Prepare for the card installation. Switch off power supply; always wear an electrostatic
discharge preventive wrist strap and use a grounding mat.
3. Select any universal slot to insert the card. Unscrew the filler bracket and remove it by pushing up the
levers on the bracket.
4. Insert the card into the guide rails of the free slot. Make sure the card's connectors make perfect contact
with the connectors on the backplane motherboard. Press down the levers of the mounting bracket to
secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
5. Now, plug in the cables supplied with the Card into the connectors on the card. Terminate the free ends of
the cables into the MDF.
Refer to the following pinout details of the card connectors for terminating the cables from the LD Port of
the card into the MDF.
For detailed instructions, refer the topics “The Main Distribution Frame (MDF)” and “Terminating Trunk and
Station Cables on the MDF”.
Also refer the topics “The Single Line Telephone Card”, “The Two-Wire Trunk Card”, “The E&M Card”,
“The Magneto Card”.
7. If you have finished installing the desired number of cards, plug in the power cord, switch ON power supply
from the mains, switch on the Power supply of the ETERNITY and observe the Reset Cycle and the LED
Pattern of the card.
Reset Cycle
Initialization
Errors
Invalid Slot detection ORANGE All LEDs Flash (250ms ON-250ms OFF) 6 times, OFF 3 sec.
Port Events:
Port Event From Master GREEN LED of the Port gets Toggled
Port Event From Module RED LED of the Port gets Toggled
The SIP-based VoIP Card enables the stations of ETERNITY to connect to the IP network and make Proxy as well
as Non-Proxy (Peer-to-Peer) VoIP calls. The card has a Registrar Server, allowing any SIP device to be registered
with it and function as an extension of the ETERNITY. With the VoIP Card, ETERNITY offers the functionality of an
IP-PBX.
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The LAN Port is used for connecting the VoIP Card to the Local Area Network to register SIP extensions through
the LAN Port.
The WAN Port is for connecting the VoIP Card to the public network over a Router/Modem. Any user on the public
network can be registered as SIP Extension through the WAN Port.
The LAN Port supports Static IP Addressing only. The WAN Port supports Static, DHCP and PPPoE IP
Addressing.
Voice Channels
There are 32 Voice Channels on the VoIP32 Card and 16 Voice Channels on the VoIP16 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
It is possible to program all 32 SIP trunks on a single VoIP Card or in a distributed manner, where more than one
VoIP card is installed in the system.
SIP Extensions
ETERNITY ME supports 999 SIP Extensions. Upto 500 SIP Extensions can be registered with a single VoIP Card.
To register more than 500 SIP Extensions, you need at least two VoIP Cards.
You can register any SIP-enabled device like an IP-phone, a Softphone, analog phone adapter, with the VoIP Card
as the 'SIP Extension' of the ETERNITY ME.
The SIP Extensions function in the same ways as other extensions of the ETERNITY. SIP Extension users can
make and receive calls from and to other extensions of ETERNITY and external numbers over PSTN, GSM, VoIP
and E&M lines26. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP
Extensions.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic “License Management”.
A SIP Extension can be registered with the VoIP Card of ETERNITY from three different locations. This helps
organizations overcome geographical distances and reduce call costs.
26. Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your coun-
try prohibit call traffic between the public telelphony networks and IP networks, you must configure Logical Partition in your system.
To know more, see “Logical Partition”.
ETERNITY ME16S
LAN Switch/Hub
LAN
IP
IP
Router
VoIP Server Card
WAN
Switch Card
Master Card
When connecting the card to the Public Network, you would require the following items/information:
• A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
If you want to make only Peer-to-Peer calls27 (calls made without the intervention of a SIP Server or Proxy
Server) you do not need the service of an ITSP.
If you have subscribed one or more SIP Accounts to make SIP calls, ask your Internet Telephony Service
Provider (ITSP) for the following information:
• SIP ID/User ID
• Authentication User ID
• Authentication Password
• SIP Registrar Server Address
• SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
• Network Information:
27. Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
ETERNITY ME16S
LAN
VoIP Server Card
WAN
Switch Card
Master Card
LAN Switch/Hub
IP
IP
Router
The card is located behind the NAT Router and Private IP is assigned to the WAN port.
When connecting the card in a Private Network, you would require the following information:
• IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
• IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
• Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
• Gateway Address
• DNS Address
• DNS Domain Name (if applicable)
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
1. Get the items/information listed ready before you install the VoIP card and connect it to the IP network.
2. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
4. Select any of the free Universal Slots of ETERNITY to insert the VoIP Card. Unscrew and remove the filler
bracket of the slot. Preserve the filler bracket for future use.
5. Insert the card into the guide rails of the slot. The card should be inserted deep enough to make perfect
contact with the connectors on the backplane.
6. Now, secure the card in its slot by pressing down the levers of the mounting bracket and fixing it in place
with the two screws provided.
7. Using the Ethernet cable supplied with the VoIP card, connect the LAN and the WAN Port to the IP
network, which may be Public Internet or a LAN, or both.
• Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end into the Router/Modem.
• Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
• Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end of the cable into the Router/Modem.
• Connect the LAN Port of the VoIP Card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all other installation tasks, you may switch on power supply and observe the Reset
Cycle and the LED indication of the VoIP Card.
LED Indication
There are two LEDs on the VoIP Card: LED 1 and LED 2.
Once Stack Construct COMMAND is received from Master, LED will Red Continuous ON
remain ON Red and wait for the WAN and DSP download response.
WAN Stack Construction Failed -(PPPoE connection is not established Red ON 1 sec-OFF 1 sec
or IP Address Invalid or DHCP Client does not retrieve network ON 1 sec-OFF 1 sec
parameters or any other reason)
DSP Image not downloaded successfully (in any DSP but not all)
DSP Image not downloaded successfully (in any DSP but not all)
SIP Trunk Status will be indicated by LED2 only after you have programmed LED Indication in VoIP Port
Parameters.
ETERNITY ME supports up to 999 SIP Extensions. The SIP Extensions function like DKP/SLT extensions of the
ETERNITY ME. SIP Extension users can make and receive calls to any extension user of the ETERNITY and to
external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP28.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY ME.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY ME, and you must have the IP8
License. For more information on Licensing, see “License Management”.
You can register upto 500 SIP Extensions with a single VoIP Card of ETERNITY ME. However, at a time, only as
many extensions as the number of “Voice Channels” supported by the “The VoIP Card” can make calls.
You can register the same SIP Extension from three different locations.
You may also connect the Standard and Extended IP Phones of Matrix.
The Matrix Extended IP Phone, SETU VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
• Background Music
• CO Call Waiting
• Hot Desking
• Live Call Screening
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
If the ETERNITY ME Master Card and VoIP Card are connected to a Public Network,
• Connect SETU VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
• Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
28. Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
“Logical Partition”.
LAN Switch/Hub
LAN
IP
IP
Router
VoIP Server Card
WAN
Switch Card
Master Card
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
• the Registrar Server Address of ETERNITY ME
• the Registrar Server Port
• the SIP ID
• Authentication ID and Password.
If the ETERNITY ME Master Card and VoIP Card are connected to a Private Network (Behind the NAT),
ETERNITY ME16S
LAN
VoIP Server Card
WAN
Switch Card
Master Card
LAN Switch/Hub
IP
IP
Router
• You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
• Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
• use DHCP option 224 and Data Type as ‘String’ to provide Server Address to the Extended IP
Phones.
• Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY ME in the DHCP option.
• Log in to Jeeves. For instructions, read the topic “Using Jeeves” under Configuring ETERNITY.
• Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see “Configuring SIP Extensions”.
• For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic “Matrix Extended IP Phone Settings” under Configuring SIP Extensions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SETU VP248. For the purpose of illustration, the premium model, SETU VP248P, has been used.
• When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
• Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
• Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
• When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
If you attach the Foot Stand at 45°, the phone will be placed in an almost upright position on your
desk.
• Decide which of these positions would work for you best and accordingly attach the Foot Stand.
• Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
• Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack on the left side panel of the phone.
OR
You may plug a headset with an RJ11 connector in to the headset port at the bottom of the phone.
5. Connect the LAN Port of SETU VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone29. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
When you power the phone, the boot process will be initiated in the following sequence.
• The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
• As soon as the ‘Loading...’ message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
• Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
• When the cursor is placed under the Extended IP Phone, press Enter key.
We l c o me to M a t ri x
L oa d in g V 0 5R 0 1 Ex t S I P
• After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See “Network Settings”at the end of
this topic.
• The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY ME.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
• On successful download of all configuration files, the phone attempts to register with ETERNITY ME.
• On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
1. During start-up, when the phone prompts you to change the network settings after loading the firmware.
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to ‘Local Menu’.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
CA03 * 0 #
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
• In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
• Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
• You can configure all network parameters described below, except the MAC Address.
Connection Type
• Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
• If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
To enter the dot/period in the IP Address, press the digit key ‘1’ twice.
Subnet Mask
• If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key ‘1’ twice.
Gateway Address
• If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
• If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
• If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
• If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key ‘1’ twice.
• If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
• If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
• This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
• If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
• ETERNITY ME Master Card works as the Auto Configuration Server for the phone. Enter the IP Address
or the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as ‘String’ to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
• Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic30.
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
• Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
• To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
• Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
• Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
• Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
• Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
• Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
• Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
30. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), i.e. better handling
of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred to as
Class of Service (CoS) which is defined by IEEE 802.1P.
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see “Using PCAP Trace for Matrix Extended IP Phone”, under PCAP Trace.
When you change the Network Settings, the phone will restart.
• In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
N E T W O R K S TAT U S
MAC: 0 0:1 b:0 9:0 0:9 a:a 7
IP : 1 92. 16 8. 2 0 1 .2 0 5
MASK: 2 5 5 . 25 5 . 2 5 5 .0
G W: 1 9 2 . 16 8 . 2 0 1 .3
DNS:
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
• S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
ME.
• S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY ME.
The VMS Card provides a full fledged, 'in-skin' Voice Mail System with auto attendant and voice mail features. It is
designed to provide a variety of voice applications that are commonly supported by any external Voice Mail
System.
Each Mailbox has the capacity of storing 254 messages. The size of each Mailbox is set by default to 5 minutes.
The maximum message length for each mailbox is set by default to 15 seconds31.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 1GB Pen Drive with the VMS
card. The Pen Drive supports 18 hours of recording.
The VMS Card has an Ethernet Port, a communication port (COM1), a USB port, and four LEDs.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or in a LAN) to access and use the
embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages. The Ethernet
Port can also be used for Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The Pen Drive provided by Matrix with the
VMS Card is connected to this port. All the voice messages, mailbox messages, greetings and other messages and
prompts are stored in Pen Drive.
The Pen Drive is factory fitted and shipped with the card.
LEDs
The ETERNITY ME VMS16 has four LEDs: L1, L2, L3 and L4.
The L1 shows the 'Status'.
31. When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of messages is 999 seconds.
2. Unpack the card and check the package contents with the packing list.
3. Switch OFF Power supply and unplug the power cable of the ETERNITY.
4. Remove the filler bracket of the empty slot you have selected for installing the VMS card, by removing the
screws and pressing up the levers of the filler bracket.
5. Insert the VMS card into the guide rails of the slot. Make sure its connectors fit perfectly into those on the
backplane.
6. Secure the card in its slot by pushing down the levers of the mounting bracket and fixing the card with the
two screws provided.
7. If you have completed all installation tasks, switch ON the ETERNITY and observe the Reset Cycle and
the LED pattern of the VMS Card.
At power on
• All LEDS are OFF.
• L1 LED glows green as soon as initialization process starts. Initialization process starts after approximately
100 seconds.
• After approximately 80 seconds of glowing of L1, LEDs will follow the sequence as under:
In normal condition
LED L1 will behave in the following manner:
Pen Drive Error - Pen Drive Full RED Five times 100ms ON-100ms OFF (for 1 sec) - 1 sec OFF
8. Open Jeeves, and configure the VMS Card. Refer the topic “Configuring the Voice Mail System (VMS)” for
further configuration instructions.
9. If you need to generate configuration and debug reports, connect the COM Port of the VMS Card with that
of a PC using the communication cable supplied with the card.
10. If you want to use FTP to upgrade Software of the VMS Card, upload recorded voice messages, store
Back-up of Configuration files and Mailbox messages, connect the Ethernet port of the VMS card to a
standalone PC or a PC on LAN.
• Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a to a LAN PC, you need to make sure that
• The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the LAN PC are not the
same.
• The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
Power ON
1. If you have completed all the installation tasks, switch on power supply.
• For PSUNI card installed in the system, connect the three-prong plug of the power cord from the
ETERNITY into the AC outlet, and switch on power supply.
• For PS48V card installed in the system, keep the MCB Switch ON and power the FCBC.
Reset Cycle
• All the LEDs of the system, the cards and the keys of the DKP attached to the System are turned on.
Interpreting LEDs
The functioning of the LEDs of the system and the various cards and their meaning are summarized in the
tables below. This will help you to verify if the system is operating properly and locate faults, where they
occur.
3. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
4. Now, close the enclosure cover, pressing down the snap lug as you push each part of the cover in its
place.
• The Matrix ETERNITY is to be installed by persons trained and experienced in telecom wiring.
• The person installing the ETERNITY must be familiar with trunks, physical wiring of the MDF on both
the exchange (PBX) side and the line side (CO).
• When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
system and its cards to prevent damage to the system and harm to yourself.
• Use a grounding mat and wear an anti-static strap/belt. Read the dos and don'ts listed in “Protecting
ETERNITY and Yourself”.
• If you have complied with the requirements and instructions described in '“Before You Start”, you may
now begin the installation of your ETERNITY GE.
The Matrix ETERNITY GE is shipped factory fitted with the Power supply card, the Master and Switch Card in their
respective fixed slots (refer the section “Know Your ETERNITY”).
The cards - BRI, T1E1PRI, GSM, VoIP, DKP, TWT, SLT, VMS, E&M - are shipped separately as per the order
placed by individual customers. These cards are installed in any of the Universal slots.
Illustrated below is the position of the fixed and universal slots in each variant of ETERNITY GE.
ETERNITY GE12S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
ETERNITY GE3S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
Follow the installation instructions for cards described here also when you expand the system (add more cards) or
remove or swap cards for maintenance and repair.
1. Unpack the box. Check the package contents (see “Packing List”). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation.
6. Check the voltage at the power point from where the supply is to be given to the system. It should be as
per the specifications. Earth the system properly. (Refer “How to Make the Telecom Earth”)
Inserting Cards
7. Make sure that the ETERNITY power is off and the power cord is unplugged.
9. Unscrew and remove the filler bracket that covers the card-slot opening of the slot you intend to use.
10. Hold the card with the connectors facing you. Do not grab the card from both ends.
11. Slide the card into the slot, along the guide rails provided for each slot at the top and bottom planes.
12. Ensure that the cards are inserted deep enough for all the connector pins on the cards make complete
contact with those of the motherboard on the backplane.
Do not force the card into the slot. Doing so can damage the card or the slot connector.
13. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
secure the card with the screw provided.
Detailed installing instructions are provided for each card - DKP, SLT, TWT, ISDN BRI, ISDN T1E1PRI,
GSM, VoIP, E&M - later in this section. Refer to them when installing each card type.
• If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
• Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
17. Using the cables supplied with each card, and terminate the cables in the Main Distribution Frame (SLT,
DKP, TWT, and E&M lines), the NT1 device (ISDN BRI lines), ISDN Modem (ISDN PRI Lines), as
applicable.
For cards with multiple RJ45 connectors, the cables are bunched together, but each cable is identified by a
distinct color marked at the Boot edge and the Insulation edge of the cable.
The color markings make it easy for you to identify the connector on the card into which the cable is
plugged in, and the hardware ports on that connector. For example, the cable marked with blue on the
Boot and Insulation edge is connected to the first port of the card, containing the hardware ports 01-04.
18. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
Two types of Power Supply Cards are supported by the Matrix ETERNITY GE models: PS UNI and PS48V.
• PS UNI Card with 100-240VAC, 47-63Hz Mains as Input AC Voltage Power Supply.
This card is designed on the SMPS scheme. As this card does not have any provision for battery backup,
it is recommended that a UPS be connected to keep the system powered during outages.
This card has four LEDs, a Mains Switch, and a Socket assembly for connecting the mains cord.
• PS48V Card with 48VDC as Input DC Power Supply Voltage. A Float cum Boost Charger (FCBC) is
required to feed 48VDC power to the card. The FCBC works on input AC mains.
The card has four LEDs, an MCB Switch, a power ON/OFF Switch, and a 3-way termination block for
connecting the power cord.
Both, the PS UNI card and the PS48V Card provide DC output voltages as: +3.5V, +5.0V, -27V and -85V.
These are indicated by LEDs.
The Power Supply Card is delivered factory fitted, when you buy the system. However, if you want to remove the
card for the purpose of maintenance or replace it with a new one, please follow the instructions below:
1. Unpack the Power Supply Card and verify the package contents.
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
5. If installing the PSUNI card, connect the three-pin power cord into the socket of the PS UNI card and plug
in the cord into the mains supply.
You may connect the PSUNI Card to a UPS to keep the system live during power outages.
Select a UPS considering the typical power consumption of ETERNITY presented in the table below.
6. If installing the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from
the FCBC output into the 3-way termination block on the PS48V card.
Polarity is critical. Ensure that the wires are connected with the correct polarity. Follow the standard color
codes used by FCBC manufacturers:
Color Signal
Red +48VDC
Black GND
Green Earth
It is recommended that you measure the voltage before connecting the power cable to the power supply
card. Ensure that the earth is connected.
48V Battery
32. When the batteries are drained, the FCBC goes into the charge mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery volt-
age goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to charge mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the bat-
teries are charged with higher current initially.
The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. If 48V/26Ah
batteries are connected to the FCBC for the ETERNITY GE system you can ensure a backup time of 2.5 to
3 hours. The FCBC uses the constant voltage charging method. So, the batteries get charged faster if less
power is consumed by the system when in mains mode.
The CPU card is designed for the ETERNITY GE models as a combination of the Master Card and Switch Card of
ETERNITY ME. The CPU card has the functions of both cards; it manages the entire system, controls all other
slave cards (SLT, DKP, TWT+SLT, DKP+SLT, E&M, BRI, VMS, T1E1, GSM, VoIP, etc.). All configuration and
programming information is stored on this card.
The CPU card occupies a fixed slot, second from the left, with a unique arrangement of connectors. So no other
card can be inserted in the slot of the CPU card.
The CPU card has an Ethernet Port, Communication Port, a Digital Input Port (DIP), a Digital Output Port (DOP),
an Analog Input Port (AIP), an Analog Output Port (AOP), and a USB Port on the front panel.
Communication Port
This is a single asynchronous, serial, full duplex RS232C communication port, labeled as COM1. The COM Port
has a DB-9 connector. The COM port is meant for connecting a PC to the ETERNITY GE. With a PC connected to
the ETERNITY GE you can install and operate from the COM Port the following features:
Ensure that the device connected to the DIP complies with the technical specifications of this port.
Refer the topic “Digital Output Port (DOP)” for more information.
The music from the external source can be played as 'Music-on-Hold' to internal as well as external callers. Refer
the sections “Connecting an External Music Source” and “Music on Hold (MOH)”.
When you connect an external paging device to the AOP, ensure that it complies with the technical specifications of
the port.
Master Ethernet Port RJ45 Facia To connect ETERNITY to a PC/LAN (to run various
software; download system activity reports)
USB port (Device Port) USB Fascia To download data on to the ETERNITY through a PC.
Communication (COM DB-9 female Fascia To connect a PC (to run various software; download
Port) system activity reports).
Digital Output Port Push-type Fascia To connect an automated application devices: hooter,
siren, door lock, fire alarm, bell, water pump, lights, etc.
LED
The CPU card has two dual color (Green and Red) LEDs.
• LED 1 - L1 works as a Heart Bit of CPU Card. In Normal Condition, L1 will be turned ON Green for 1 sec
and OFF for 1 sec.
• LED 2 - L2 indicates the Layer Application status. In Normal condition, L2 will be turned on Orange and will
blink very fast.
Case 1: L1 is steady GREEN/OFF and LED2 is OFF. This means the Application has got hanged and
there is some problem at Application side code.
Case 2: LED1 is steady GREEN/OFF and LED2 is GREEN/RED/ORANGE. This means the Layer has got
hanged and there is some problem with the Layer side code.
BC (default) Normal.
J5 AB External Music.
BC Internal Boot.
BC JTAG Mode.
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the CPU Card into the guide rails of the slot designated for the card. On all variants of ETERNITY
GE, the second slot from the left (next to the Power Supply Card) is designated for the CPU Card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
3. Connect the Ethernet Port of ETERNITY with the Ethernet Port of the stand-alone PC using the Ethernet
cable supplied with ETERNITY.
You need to connect to the PC via the Ethernet port for the following functions:
4. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
any standard Communication Cable.
You need to connect to the PC via the COM Port for the following functions:
• access the web-based programming tool Jeeves from any PC on the LAN.
• set up and run software applications such as PMS and CAS on any PC on the LAN.
• generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY GE to a to a standalone/LAN PC, you need to make sure that
• The IP Address of the Master Ethernet Port of the ETERNITY GE and the Ethernet Port of the PC do
not conflict.
• The Master Ethernet Port of ETERNITY GE and the Ethernet Port of the PC are in the same Subnet.
If the system is connected to a LAN PC, ask the LAN Administrator to assign an IP Address and a Subnet
Mask to the ETERNITY GE35.
• Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
You get programming tone.
To change IP Address
9. Switch off power supply and continue with other installation tasks.
Use 0.5mm, non-stranded cables to connect the sensor device to the DIP.
• strip off about half a centimeter of the insulation off the wire ends of the sensor device.
• using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
• insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
36. If the ETERNITY is connected to a LAN (without a DHCP server), use the IP Address and Subnet Mask given by your LAN Admin-
istrator as the new IP Address, Subnet Mask.
• A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device can be
interfaced with the DOP via this DC contactor.
• The DOP has a two-wire, push-in (spring clamp action) connector to attach the relay device.
Operation Time 5 ms
• strip off about half a centimeter of the insulation off the wire ends of the gadget.
• using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
• insert the stripped ends of the two wires into the two (green-color) openings of the connector, with
one wire in each opening.
• ensure that both wires fit snugly into the openings.
• release pressure on the levers. Both wires will be held in place by spring clamp action.
Connect a good quality external amplifier and matching speakers to the port.
Specification Value
Use shielded cable for connecting the amplifier with the speakers.
Specification Value
• strip off about half a centimeter of insulation of the wire-pair of the external music device.
• using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the AIP connector.
• insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
• ensure that both wires fit snugly into the openings.
• release pressure on the levers. Both wires will be held in place by spring clamp action.
Also refer the topics “Music on Hold (MOH)”, “Background Music (BGM)”, “External Music”.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
The Single Line Telephone (SLT) Card provides the interface to connect as extension phones, any standard, two-
wire, analog single line telephone instrument - rotary, pulse-tone, cordless, feature phones with or without Calling
Line Identification.
The SLT Card is available in the following configurations for the models of ETERNITY GE. SLT interface also is
available in combination with Two-wire trunk and digital key phone interfaces on a single card.
ETERNITY GE Card DKP4+SLT16 Combination card, with 4-ports to connect to 4 Digital Key Phones and
16 ports to connect 16 Single Line Telephones
ETERNITY GE Card Combination card, with 2 ports to connect 2 Two-wire Trunk lines, 2
TWT2+DKP2+SLT16 ports to connect 2 Digital Key Phones, and 16 ports to connect 16
Single Line Telephones
ETERNITY GE Card TWT8+SLT8 Combination card with 8 ports to connect 8 Two-wire Trunk lines, and
8 ports to connect 8 Single Line Telephones
ETERNITY GE Card TWT4+SLT16 Combination card with 4 ports to connect 4 Two-wire Trunk lines, and
16 ports to connect 16 Single Line Telephones
The maximum number of SLT ports supported by the variants of ETERNITY GE are:
Connectors
The SLT Cards have RJ45 connectors, with each connector having 4 SLT ports. A multi-pair, MDF cable is supplied
for each connector.
LEDs
The Card SLT8 has 2 LEDs, while SLT20 has no LED:
• The LEDs indicate the health of the card during the Reset Cycle.
• the status of any one of the station ports during normal functioning of the system.
You can monitor any of the SLT Station ports by assigning the LED to that port37.
37. To do this, enter SE mode, enter the programming mode from any station connected to the ETERNITY, by dialing 1#91-1234. Dial
the command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is
the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the
card, which will monitor the port. Exit programming mode by dialing '00'.
a. The current LED state will remain the same until the next command is received from the ap-
plication on the SLT Port. For example, if the current LED state is Green/Red ON, on the
next command received, the LED will be turned OFF. It will remain OFF until the next com-
mand is received. When the next command is received it will be turned Green/Red ON
again. This process continues.
b. Same as above note.
1. Decide the number of SLT extensions required and arrange for as many telephone instruments.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require
you to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of
the phone to dial Flash.
2. Unpack the SLT card and check the package contents. Ensure that the power supply is switched off,
before you begin the installation of the card. Always wear an electrostatic discharge prevention wrist strap/
belt and use a grounding mat.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket! You may require it at a later stage.
4. Insert the SLT Card into the guide rails of the free slot you selected for the card.
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
bracket with the two screws provided.
If you are installing more than one SLT card, you can install the second card in any other free slot. It is not
necessary to install the second/third card in the subsequent slots.
6.Use the cables supplied with the SLT card to connect the SLT wires with the Main Distribution Frame.
For each connector on the SLT Card, there is a separate 4-pair cable with an RJ45 jack on one end and
free at the other end.
The color markings make it easy to identify the connector in which the cable is plugged in and the ports on
the connector.
Refer the illustrations below to help you identify which cable to plug into each connector and the ports on
each connector.
7. Plug in the RJ45 end of the MDF cables supplied with the card into the respective connectors with the help
of the color markings on the cables as illustrated above for each SLT Card type.
8. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the “The Main Distribution Frame (MDF)”.
Each wire-pair from the ETERNITY SLT Port must be terminated to the bottom of the Krone Connector,
while the wire-pair of the extension line to be connected to this port must be terminated on the top of the
Krone connector. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
10. Connect the SLT instruments you have arranged for. Plug in the SLTs into the wall socket/outlets.
• For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.
• When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the SLT will be
connected on the first port on the connector.
For the Building Intercom application, ETERNITY GE supports the Intercom Line Card (ILC).38
You can connect any standard, two-wire, analog single line telephone instrument - rotary, pulse-tone, cordless,
feature phones with or without Calling Line Identification to the Intercom Line card.
The Intercom Line Card is available in the following configurations for the variants of ETERNITY GE.
Choose an ILC Card with the configuration that meets your requirement for intercom ports. Also, consider the
maximum Port capacity of the system you are installing. The maximum number of intercom ports supported by the
variants of ETERNITY GE are:
Connectors
The ILC Cards have RJ45 connectors, with each connector having 4 ports. A multi-pair, MDF cable is supplied for
each connector.
LEDs
The Card ILC8 has 2 LEDs, while ILC 20 has no LED:
• The LEDs indicate the health of the card during the Reset Cycle.
• the status of any one of the station ports during normal functioning of the system.
1. Decide the number of intercom extensions required and arrange for as many telephone instruments.
2. Ensure that the extension wiring is completed according to your requirements. The extension cables from
the wall jack are terminated in the Main Distribution Frame and the telephones are connected to the wall
jacks.
3. Always wear an electrostatic discharge prevention wrist strap/belt and use a grounding mat to prevent
damage to the components of the card.
38. Check Availability. This card is supported by Firmware Version 10.06 and later only.
5. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Keep the filler
bracket for future use.
6. Insert the ILC card into the guide rails of the free slot you selected for the card.
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
7. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
bracket with the two screws provided.
9. Now, use the cables supplied with the ILC card to connect the card to the Main Distribution Frame to which
the intercom phones are connected.
For each connector on the card, there is a separate 4-pair cable with an RJ45 jack on one end and free at
the other end. Refer the illustrations of the pinout of the intercom cards to connect the wires.
10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle.
The Digital Key Phone (DKP) Card provides the interface to connect the proprietary digital key phones, EON, the
PC-based phone EONSOFT and the Direct Station Selection (DSS) Consoles with the ETERNITY.
The DKP Card is available in the following configurations for the models of ETERNITY GE.
ETERNITY GE Card Combination card, with 4-ports to connect to 4 Digital Key Phones and 16 ports to
DKP4+SLT16 connect 16 Single Line Telephones
ETERNITY GE Card Combination card, with 2 ports to connect 2 Two-wire Trunk lines, 2 ports to connect 2
TWT2+DKP2+SLT16 Digital Key Phones, and 16 ports to connect 16 Single Line Telephones
The maximum number of DKP Ports supported by each variant of ETERNITY GE is:
Connectors
The DKP Cards have RJ45 connectors, with each connector having 4 DKP ports. A multi-pair MDF cable is
supplied for each connector on the card.
LEDs
The DKP16 and DKP8 cards have two dual color LEDs:
• LED1 indicates the health of the card during the Reset Cycle.
• LED2 monitors the status of any one of the station ports during normal functioning of the system.
LED 2 can be assigned to any DKP port to monitor the status of that port39.
39. You can do this from the SE mode, by dialing the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the uni-
versal slot in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its function-
ing. LED Number is the number of the LED on the card, which will monitor the port.
a. The current LED state will remain the same until the next command is received
from the application on the DKP Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command
is received it will be turned Green/Red ON again. This process continues.
1. Decide the number of DKP extensions and DSS Consoles required and arrange for as many EON,
EONSOFT and DSS Consoles.
2. Unpack the DKP card and check the package contents. Make sure that power supply is switched off and
you are wearing an antistatic-wrist strap/belt and have a grounding mat, before handling the card.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
4. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
with the two screws provided.
If you are installing more than one DKP card, it is not necessary to install the next card in the subsequent
slot.
6. Using the MDF Cables supplied with the DKP card connect the DKP wire pairs to the Main Distribution
Frame.
For each connector on the DKP card, there is a separate cable with an RJ45 jack on one end and free wire
on the other end. The cables are bunched together. Each cable is identified by a distinct color marked at
the Boot Edge and the Insulation edge of the cable
The color markings make it easy to identify the connector and the ports to which the cable is connected.
Refer the illustrations below to help you identify which cable to plug into each connector and the ports on
each connector.
7. Plug in the RJ45 end of the DKP cables into the respective connectors guided by the color markings on the
cables as illustrated above for each DKP Card Type.
8. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY GE DKP Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the extension line to be connected to this port must be terminated on the
top of the Krone connector. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
Installing EON
Matrix offers EON, the proprietary digital key phone. EON is available in the following models:
• EON42
EON42
• To mount EON42 on a wall, detach the Foot Stand on the bottom of the phone, by pressing the snap fits of
the foot stand backwards, and lifting it from its anchorage in the mounting holes.
• Now, insert the snap fits of the foot stand into the Wall Mount bracket slots on the bottom of the phone in
the " wall up" direction.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON42. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
• Connect the handset of the EON42 to the phone body using the spring cord.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
• When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
• Connect the handset of the EON48 to the phone body using the spring cord.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
You can also plug in a headset with RJ11 connector into the Headset port at the bottom of the phone.
3. Plug one end of the RJ45 cable supplied with the phone into the RJ45 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
5. After successful handshaking and reset cycle, if the DKP Parameters have been programmed, the LCD
display of the EON will show the station number and the station name in a line. The day, date and time,
time zone in the other line.
202 Reception
M on 2 4 A U G 1 2 : 0 0
6. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, “Digital Key Phone-
Operation”.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
2. Place the DSS Console next to the DKP, EON, to which it is to be attached.
You can install two DSS consoles to a DKP. Refer “Direct Station Selection Console” for possible
combinations for installing the various models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated into the bottom of
the Krone Connector (of 'Station Lines' on the MDF).
Refer the topic “The Main Distribution Frame (MDF)” for illustration.
6. ETERNITY automatically assigns the first DSS Console discovered on the system to the first DKP, the
second DSS to the second DKP.
7. Only when two DSS Consoles are to be assigned to a single DKP, manual assignment of DSS to the DKP
is required. Refer “Configuring DKP Extensions”.
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
the speaker connectors into the dongle.
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
communication cable into the COM port of the computer.
4. Connect a wire-pair of a DKP port of the ETERNITY to the RJ11 port marked 'DKP' on the dongle.
5. Switch ON the computer. The computer must have Windows Operating System installed on it.
6. Now insert the EONSOFT CD-ROM supplied with this PC-based DKP into the CD drive of your Computer.
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
11. Click the shortcut to open the program. The EONSOFT window will open:
12. Click 'Options' at the top left of the window. A drop down menu will appear.
14. Select the COM Port to which the communication cable is connected.
This screen will appear only if the DKP port to which the EONSOFT is connected has been programmed
for parameters like Name, Station number, Date and Time.
• If this dialog box does not appear on the screen in response to the click the COM Port Option, test the
COM Port for data transfer.
• If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
• Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
• Click the button labeled 'Start Test' in the COM Port Settings dialog box.
• After clicking this button, observe the Test Result section on the dialog box.
The above screen shows that the COM Port/communication cable is working.
• If the 'Error Count' shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
• Remove the communication cable from the COM Port of the PC.
• Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
• Now, if the error count is zero, please check the Communication Cable.
• If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
The Two-Wire Trunk (TWT) Card provides the interface to connect the ETERNITY with the POTS Network. The
TWT Card supports the different standards and features of POTS Networks across the world.
The TWT Card is available in the following configurations for the variants of ETERNITY GE. TWT interface is also
available in combination with SLT and DKP ports on a single card.
ETERNITY GE Card TWT16 16-port card to connect 16 Two-wire Trunk lines from the CO network
ETERNITY GE Card TWT8 8-port card to connect 8 Two-wire Trunk lines from the CO network
ETERNITY GE Card Combination card, with 8 TWT ports to connect 8 TWT analog trunk lines and 8
TWT8+SLT8 SLT ports to connect 8 Single Line Telephones
ETERNITY GE Card Combination card, with 4 TWT ports to connect 4 TWT analog trunk lines and 16
TWT4+SLT16 SLT ports to connect 16 Single Line Telephones
ETERNITY GE Card Combination card, with 2 TWT ports to connect 2 Two-wire Trunk lines, 2 DKP
TWT2+DKP2+SLT16 ports to connect 2 Digital Key Phones, and 16 SLT ports to connect 16 Single
Line Telephones
The maximum TWT Trunk Ports supported by the variants of ETERNITY GE is:
• ETERNITY GE 3S: 48
• ETERNITY GE6S: 96
• ETERNITY GE12S: 128
Connectors
The TWT Card has RJ45 connectors, with 4 TWT ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
LED
The TWT16 and TWT8 Cards have two LEDs to indicate:
You can assign the LED to any TWT port on the card which you want to monitor40.
40. To assign the LED to a selected port for monitoring its functioning, you must enter SE mode and dial the SE Command 7902-Slot-
-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor
the port.
a. The current LED state will remain the same until the next command is received
from the application on the TWT Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command
is received it will be turned Green/Red ON again. This process continues.
1. Take all the necessary precautions prescribed for handling the cards and electronic equipment. Make sure
that power supply is turned off before you begin the installation of the card. Put on an electrostatic-
discharge preventive wrist strap/belt and use a grounding mat.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Preserve the filler bracket for future use!
4. Insert the TWT Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
5. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one TWT Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
6. Use the cables supplied for each connector on the TWT card to connect the Trunk Lines with the Main
Distribution Frame.
The cable for each connector is identified by a distinct color marked at the Boot edge and the Insulation
edge of the cable. This is to help you identify the connector and the ports to which the cable is connected.
You may refer the illustrations below to identify which cable to plug into each connector with the help of the
color markings on the cable.
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors referring to the color
markings on the cables as illustrated above for each TWT Card type.
8. Terminate the free end of the TWT Card cable into the punch down blocks of the Krone modules
designated for 'Trunk Lines' on “The Main Distribution Frame (MDF)”.
Trunk cables from the ETERNITY are to be connected with the Trunk Lines from the PSTN/CO terminated
on the MDF. Each wire-pair form the ETERNITY GE TWT Port must be terminated on the bottom of the
Krone Connector, while the wire-pair of the trunk line from the CO Network to be connected to this port
must be terminated on the top of the Krone Connector.
Refer the topics “The Main Distribution Frame (MDF)” and “Terminating Trunk and Station Cables on the
MDF”.
The BRI card provides the interface to connect ETERNITY with ISDN BRI Lines. The BRI lines may be from a
public ISDN exchange, a private ISDN exchange.
The BRI Card is available in the following configuration for the variants of ETERNITY GE.
ETERNITY GE Card BRI4 4-Port card to connect 4 ISDN BRI Lines or ISDN Compatible Devices
The maximum number of BRI lines supported by each variant of ETERNITY GE are:
Connectors
The BRI card has 4 RJ45 Connectors. A separate cable is supplied for each connector.
ISDN
Network NT 1 BRI Port
ETERNITY
Power
U-Interface S/T
(2-wire) Interface
Customer Premises
Where,
• U Interface = between the NT1 equipment and the ISDN central office.
• S/T Interface = between the ISDN user equipment, in this case, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the ETERNITY.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
• the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
• the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
Point-to-Point Configuration
BRI Line
NT BRI Port
ISDN (TE Mode)
Network
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be up to 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
ISDN NT
Network BRI Bus Bar
Terminal
BRI Port Resistance 100
(TE Mode)
Terminal
Resistance 100
ETERNITY ISDN Phone ISDN Phone ISDN Phone
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
• A maximum of 8 TEs or ISDN devices can be connected to a single NT on a bus up to 200 meters from the
NT.
• 100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
• Using this configuration, any subscriber from ETERNITY can access a BRI line and can make outgoing
calls. At the same time, another subscriber from ETERNITY or any ISDN phone shown in the figure can
make outgoing call from the same BRI. In the same way, incoming calls are possible on the same BRI.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d < 1 Km
d1 < 30 meters
ISDN NT
Network BRI Bus Bar
Terminal
Terminal BRI Port Resistance 100
(TE Mode)
Resistance 100
• You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
• However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
• Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
• If the BRI Port is to be configured in the NT mode, all the related Jumpers of each port should be set in BC
position.
• When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
• When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
BRI Line NT
Network
• Last TE equipment
• Last point of the bus bar where the last TE equipment is connected.
• If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
• Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Jumper Position for BRI Port1 Jumper Position for BRI Port2
Function
J3 J4 J3 J4
Jumper Position for BRI Port3 Jumper Position for BRI Port4
Function
J3 J4 J3 J4
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position)
7. To remove the 100 termination from the BRI port set the Jumpers J3 and J4 (provided on the BRI
module) in BC position.
1 RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1 1 RJ45 Connector
ports on BRI Bus
Bar to which the
3 3
ISDN TE
4 4 Equipment is
connected
5 5
6 6
8 8
• The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
• Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
• Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
• Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
To do this, you must change the position of the Jumpers J1 and J2 on the BRI modules (daughterboard),
on the BRI Card.
J1 J2 J1 J2 J1 J2 J1 J2
No power to be fed to the terminal Open Open Open Open Open Open Open Open
equipment
• The maximum power that can be fed to a single BRI port is 50mA.
• From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
• The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
10. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
11. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
The following diagram shows how to connect a BRI Line to the ETERNITY GE BRI port (in the TE mode).
NT-1 ETERNITY
3
Rx1 TxA
4 4
Tx Tx1 RxA
ISDN
5
Network Rx Tx2 5 RxB
6
Rx2 TxB
NC V-
Power NC V+
• V- and V+ are used when a TE is connected to BRI port (in this case the port functions as network or
NT).
13. To connect the BRI Lines to the BRI ports, refer the configuration and pinout details given below for
guidance.
4 Tx
5 Rx
3 Rx1
4 Tx1
5 Tx2
6 Rx2
3 Green-White TxA
4 Blue RxA
5 Blue-White RxB
6 Green TxB
7 Brown-White V-
8 Brown V+
3 Green-White RxA
4 Blue TxA
5 Blue-White TxB
6 Green RxB
7 Brown-White V-
8 Brown V+
14. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle and the LED pattern of the BRI Card.
The LEDs show the Status of the Ports as summarized in the table below:
The ETERNITY T1E1PRI Card provides the interface to connect ETERNITY GE to ISDN PRI Network.
When connected to T1 carrier lines, the Card supports the following signaling types:
• PRI
• Robbed Bit Signaling
• Q-Signaling (QSIG)
• E&M
When connected to E1 carrier lines, the card supports the following signaling types:
• PRI
• Channel Associated Signaling (CAS)
• Q-Signaling (QSIG)
• E&M
The T1E1PRI Card is available in the following configuration for ETERNITY GE:
ETERNITY GE Card T1E1PRI 1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Single Compatible Device
The maximum number of PRI Lines supported by each variant of ETERNITY GE is:
Connectors
The T1E1PRI card has an RJ45 Connector. A cable with RJ45 plugs on both ends is supplied for the connector.
LEDs
The ETERNITY GE T1E1PRI Card has 2 LEDs - L1 and L2 - for indicating the port states.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket.
5. Insert the T1E1PRI Card into the guide rails of the free slot you selected for the card. Make sure that the
connectors on the card make perfect contact with those of the slot on the backplane motherboard.
6. Now, press down the levers on the card mounting brackets to secure the card in its slot. Fix the card in
place with the two screws provided.
Customer Premises
ETERNITY
ISDN G.703
Modem
Network 4-wire HDSL DTE
4-wire
PRI Port
(RJ-45 Connector) (RJ-45 Connector)
G.703
Modem
Power
• Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line—one at
the Local Exchange and other at the Customer's Premises.
• At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
• The DTE interface of the modem is to be connected to the PRI port (RJ-45 connector on the Matrix
ETERNITY GE T1E1PRI Card).
8. Plug in one end of the RJ45 cable supplied with the card into the card's connector. Plug the other end of
the RJ45 cable into the Network Termination Unit.
9. Refer the following pin details for connecting the Network Termination Unit with the ETERNITY.
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)
5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the installation guide of the HDSL Network Termination Unit being used by you.
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
10. Repeat the same steps to install another card. It is not necessary to install the other T1E1PRI cards in a
sequence. Any card can be installed in any of the slots.
11. If you have completed all other installation tasks. Power the system. After the Reset Cycle, observe the
LED patterns of the T1E1PRI Card.
LED Patterns
The ETERNITY GE T1E1PRI Card has 2 LEDs: L1 and L2. Given below are the LED Patterns defined for each port
state in the different signaling types supported by the ETERNITY GE.
LED1 Pattern:
LED2 Pattern:
LED1 Pattern:
LED1 Pattern:
LED2 Pattern:
LED2 Pattern:
Near end loop back wait Before activate RED 100ms ON-100 ms OFF
Near end loop back wait Before deactivate RED 500ms ON-500 ms OFF
Far end loop back wait after activate GREEN 100ms ON-100 ms OFF
Far end loop back wait after deactivate GREEN 500ms ON-500 ms OFF
LED2 Pattern:
The Mobile Card interfaces the ETERNITY with GSM/3G networks. It routes calls made and received over mobile
networks, like a mobile handset.
The mobile card does not support GPRS features, Fax and Data services, network supported services,
except CLIR and USSD.
The Mobile card is available in the following configurations for ETERNITY GE.
ETERNITY GE Card GSM4 4-port card to connect to 4 GSM networks (4 SIM Cards can be installed)
ETERNITY GE Card GSM4 3G 4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards can
be installed)
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
• ETERNITY GE3S: 12
• ETERNITY GE6S: 24
• ETERNITY GE12S: 32
Antenna
ETERNITY GE Card GSM4 has a single antenna for the four ports. A splitter connects all the four ports on the card
into a single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another
position (in case of weak signal).
2. Make sure that the ETERNITY is installed at a location where sufficient network coverage is available. The
power supply should be turned off, and you must be wearing an electrostatic discharge preventive wrist
strap and a have a grounding mat, before you begin handling the card.
5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same GSM service provider or of different service providers.
6. Insert the Mobile card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
10. Wait for the system to register with the Mobile network. By default, the Mobile ports are set to select and
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.
• At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
• Each time the Mobile Port sends a request, such as Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
The E&M Card of the ETERNITY provides the interface for analog trunking to connect various communication
equipment telephone switches, Routers, Leased Lines, etc. using Tie-Lines.
• Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer “PLCC-An Introduction” to know more.
• “Closed User Group (CUG)”, where several PBXs are connected with each other through E&M tie lines41.
• PBX expansion, where two PBXs are connected with each other with E&M tie lines.
An E&M Port can be programmed to behave as a Trunk Interface, a Subscriber (Station) Interface or both, as a Tie
Line with the dual personality of a Trunk and a Subscriber.
The ETERNITY E&M Card is available in the following configuration for the ETERNITY GE
The number of E&M lines that you can interface with the ETERNITY using the E&M Card varies according to
number of E&M ports supported by the each variant of ETERNITY GE.
The maximum number of E&M ports supported by each variant of ETERNITY are:
• ETERNITY GE3S: 12
• ETERNITY GE6S: 24
• ETERNITY GE12S: 48
41. The PBXs in a “Closed User Group (CUG)” can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
42. This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
LEDs
The ETERNITY GE E&M4 Card has 4 LEDs to indicate the functioning of the ports.
OR
• a Trunk - works like a trunk interface when any of the stations of the PBX makes an outgoing call through
it.
OR
• a Tie Line - takes on a dual personality: functioning as both a station and a trunk. The E&M port works like
a station interface for incoming calls. It works like a trunk interface when any station makes an outgoing
call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
“PLCC-An Introduction” to know more.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
• To select the Type-V connection for the E&M Port, set Jumpers J1 and J2 (located on the E&M
module) in BC Position.
• By default all the E&M Ports are set to support 2-wire Speech Interface.
• To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
• To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
5. Now, select a free slot for the E&M card. Unscrew and remove the filler bracket by pushing up the levers
on the bracket. Preserve the filler bracket for future use.
6. Insert the E&M Card into guide rails of the empty slot. Make sure the connectors on the card make perfect
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M card into the RJ45 connectors on the E&M Card.
8. Connect the free ends of the cables into the E&M Ports of the other PBX/Router/Tie Line equipment by
appropriate crossing of the wires.
Refer the following pin-out details for the E&M Card and for each Interface and Speech Interface Type.
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the PBX
when speech is established. The E&M Card supports this facility. The ETERNITY sends CCS signal to the
PLCC panel.
• When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
• When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
10. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the LED
pattern of the E&M Card.
After 60-90 seconds RED L1, L2, L3, L4 ON 500ms - L1, L2, L3, L4 OFF
After 65-95 seconds RED L1, L2 L3, L4 ON 500ms - L1, L2, L3, L4 OFF
The SIP-based VoIP Card enables the stations of ETERNITY to connect to the IP network and make Proxy as well
as Non-Proxy (Peer-to-Peer) VoIP calls. The card has a Registrar Server, allowing any SIP device to be registered
with it and function as an extension of the ETERNITY. With the VoIP Card, ETERNITY offers the functionality of an
IP-PBX.
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The VoIP card is available in the following configuration for the ETERNITY GE.
The LAN Port is used for connecting the VoIP Card to the Local Area Network to register SIP extensions through
the LAN Port.
The WAN Port is for connecting the VoIP Card to the public network over a Router/Modem. Any user on the public
network can be registered as SIP Extension through the WAN Port.
The LAN Port supports Static IP Addressing only. The WAN Port supports Static, DHCP and PPPoE IP
Addressing.
Voice Channels
There are 32 Voice Channels on the VoIP32 Card and 16 Voice Channels on the VoIP16 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
It is possible to program all 16 SIP trunks on a single VoIP Card or program them in a distributed manner, where
more than one VoIP card is installed in the system.
SIP Extensions
ETERNITY GE supports 500 SIP Extensions. Any SIP-enabled device like an IP-phone, a Softphone, analog
phone adapter, can be registered with the VoIP Card and function as the 'SIP Extension' of the ETERNITY GE.
The SIP Extensions function in the same ways as other extensions of the ETERNITY. SIP Extension users can
make and receive calls from and to other extensions of ETERNITY and external numbers over PSTN, GSM, VoIP
and E&M lines43. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP
Extensions.
A SIP Extension can be registered with the ETERNITY GE from three different locations. This helps organizations
overcome geographical distances and reduce call costs.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic “License Management”.
ETERNITY GE12S
LAN Switch/Hub
LAN
IP
IP
VoIP Server Card
WAN Router
CPU Card
43. Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your coun-
try prohibit call traffic between the public telelphony networks and IP networks, you must configure Logical Partition in your system.
To know more, see “Logical Partition”.
• A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
If you want to make only Peer-to-Peer calls44 (calls made without the intervention of a SIP Server or Proxy
Server) you do not need the service of an ITSP.
If you have subscribed one or more SIP Accounts to make SIP calls, ask your Internet Telephony Service
Provider (ITSP) for the following information:
• SIP ID/User ID
• Authentication User ID
• Authentication Password
• SIP Registrar Server Address
• SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
• Network Information:
ETERNITY GE12S
LAN
LAN Switch/Hub
IP
IP
VoIP Server Card
WAN
Router
CPU Card
44. Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
• IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
• IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
• Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
• Gateway Address
• DNS Address
• DNS Domain Name (if applicable)
VoIP Card connected to the Public Network for Matrix Extended IP Phones
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
1. Get the items/information listed ready before you install the VoIP card and connect it to the IP network.
2. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
4. Select any of the free Universal Slots of ETERNITY to insert the VoIP Card. Unscrew and remove the filler
bracket of the slot. Preserve the filler bracket for future use.
5. Insert the card into the guide rails of the slot. The card should be inserted deep enough to make perfect
contact with the connectors on the backplane.
6. Now, secure the card in its slot by pressing down the levers of the mounting bracket and fixing it in place
with the two screws provided.
• Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end into the Router/Modem.
• Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
• Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end of the cable into the Router/Modem.
• Connect the LAN Port of the VoIP Card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all other installation tasks, you may switch on power supply and observe the Reset
Cycle and the LED indication of the VoIP Card.
LED Pattern
There are two LEDs on the VoIP Card: LED 1 and LED 2.
• LED 2 indicates the status of any of the SIP Trunks to which this LED is assigned.
Once Stack Construct COMMAND is received from Master, LED will Red Continuous ON
remain ON Red and wait for the WAN and DSP download response.
WAN Stack Construction Failed -(PPPoE connection is not established Red ON 1 sec-OFF 1 sec
or IP Address Invalid or DHCP Client does not retrieve network ON 1 sec-OFF 1 sec
parameters or any other reason)
DSP Image not downloaded successfully (in any DSP but not all)
DSP Image not downloaded successfully (in any DSP but not all)
SIP Trunk Status will be indicated by LED2 only after you have programmed the LED Indication in the VoIP
Port Parameters.
ETERNITY GE supports up to 500 SIP Extensions. The SIP Extensions function in the same way as DKP/SLT
extensions of the ETERNITY GE. SIP Extension users can make and receive calls to any extension user of the
ETERNITY and to external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP45.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY GE.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY GE, and and must have the IP8
License. For more information on Licensing, see “License Management”.
You can register upto 500 SIP Extensions with a single VoIP Card of ETERNITY GE. However, at a time, only as
many extensions as the number of Voice Channels supported by the VoIP Card can make calls.
You can register the same SIP Extension from three different locations.
You may connect the Standard and Extended IP Phones of Matrix as SIP Extensions.
The Matrix Extended IP Phone, SETU VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
• Background Music
• CO Call Waiting
• Hot Desking
• Live Call Screening
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
If the ETERNITY GE Master Ethernet Port and the VoIP Card are connected to a Public Network,
• Connect SETU VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
• Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
45. Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
“Logical Partition”.
LAN Switch/Hub
LAN
IP
IP
VoIP Server Card
WAN Router
CPU Card
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
• the Registrar Server Address of ETERNITY GE
• the Registrar Server Port
• the SIP ID
• Authentication ID and Password.
If the ETERNITY GE Master Ethernet Port and VoIP Card are connected to a Private Network (Behind the NAT),
ETERNITY GE12S
LAN
LAN Switch/Hub
IP
IP
VoIP Server Card
WAN
Router
CPU Card
• Connect SETU VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
• Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
• use DHCP option 224 and Data Type as ‘String’ to provide Server Address to the Extended IP
Phones.
• Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY GE in the DHCP option.
• Log in to Jeeves. For instructions, read the topic “Using Jeeves” under Configuring ETERNITY.
• Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see “Configuring SIP Extensions”.
• For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic “Matrix Extended IP Phone Settings” under Configuring SIP Extensions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SETU VP248. For the purpose of illustration, the premium model, SETU VP248P, has been used.
• When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
• Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
• Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
• When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
If you attach the Foot Stand at 45°, the phone will be placed in an almost upright position on your
desk.
• Decide which of these positions would work for you best and accordingly attach the Foot Stand.
• Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
• Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack on the left side panel of the phone.
OR
You may plug a headset with an RJ11 connector in to the headset port at the bottom of the phone.
5. Connect the LAN Port of SETU VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone46. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
• All keys with LED, including the Speaker key, and the Ringer LED, will glow.
• The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
• As soon as the ‘Loading...’ message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
• Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
• When the cursor is placed under the Extended IP Phone, press Enter key.
We l c o me to M a t ri x
L oading V 05R01 Ext SI P
• After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See “Network Settings”at the end of
this topic.
• The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY GE.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
• On successful download of all configuration files, the phone attempts to register with ETERNITY GEGE.
• On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
1. During start-up, when the phone prompts you to change the network settings after loading the firmware.
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to ‘Local Menu’.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
CA03 * 0 #
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
• In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
• Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
• You can configure all network parameters described below, except the MAC Address.
Connection Type
• Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
• If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
To enter the dot/period in the IP Address, press the digit key ‘1’ twice.
Subnet Mask
• If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key ‘1’ twice.
Gateway Address
• If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
• If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
• If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
• If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key ‘1’ twice.
• If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
• If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
• This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
• If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
• ETERNITY GE CPU Card works as the Auto Configuration Server for the phone. Enter the IP Address or
the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as ‘String’ to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
• Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic47.
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
• Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
• To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
• Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
• Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
• Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
• Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
• Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
• Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
47. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), i.e. better handling
of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred to as
Class of Service (CoS) which is defined by IEEE 802.1P.
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see “Using PCAP Trace for Matrix Extended IP Phone”, under PCAP Trace.
When you change the Network Settings, the phone will restart.
• In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
N E T W O R K S TAT U S
MAC: 0 0:1 b:0 9:0 0:9 a:a 7
IP: 1 9 2 . 1 6 8 . 2 0 1 .2 0 5
MASK: 2 5 5 . 25 5 . 2 5 5 .0
G W: 1 9 2 . 16 8 . 2 0 1 .3
DNS:
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
• S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
GE.
• S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY GE.
The VMS Card provides a full fledged, 'in-skin' Voice Mail System with auto attendant and voice mail features. It is
designed to provide a variety of voice applications that are commonly supported by any external Voice Mail
System.
ETERNITY GE Card VMS16 Voice Mail System providing voice mail facility to all extensions of ETERNITY.
Each Mailbox has the capacity of storing 254 Messages. The size of each Mailbox is set by default to 5 minutes.
The maximum message length for each mailbox is set by default to 15 seconds48.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 1GB Pen Drive with the VMS
card. The Pen Drive supports 18 hours of recording.
The VMS Card has an Ethernet Port, a communication port, a USB port, and four LEDs.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or in a LAN) to access and use the
embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages. The Ethernet
Port can also be used for Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The Pen Drive provided by Matrix with the
VMS Card is connected to this port. All the voice messages, mailbox messages, greetings and other messages and
prompts are stored in Pen Drive.
The Pen Drive is factory fitted and shipped with the card.
LEDs
The ETERNITY GE VMS16 has two LEDs: L1 and L2.
The L1 shows the 'Status'.
2. Unpack the card and check the package contents with the packing list.
48. When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of Messages is 999 seconds.
4. Remove the filler bracket of the empty slot you have selected for installing the VMS card, by removing the
screws and pressing up the levers of the filler bracket.
5. Insert the VMS card into the guide rails of the slot. Make sure its connectors fit perfectly into those on the
backplane.
6. Secure the card in its slot by pushing down the levers of the mounting bracket and fixing the card with the
two screws provided.
Connecting to a Computer
7. Now, connect the card to a standalone PC/LAN.
• Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
• Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a to a standalone/LAN PC, you need to make sure that
• The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the PC do not conflict.
• The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
At power on
• All LEDS are OFF.
• L1 LED glows green as soon as initialization process starts. Initialization process starts after approximately
100 seconds.
• After approximately 80 seconds of glowing of L1, LEDs will follow the sequence as under:
• Glow Green-500ms - Glow Red -500ms - Glow Orange 500 ms - LEDs turned OFF.
In normal condition
LED L1 will behave in the following manner:
Pen Drive Error-Pen Drive Full RED Five times 100ms ON-100ms OFF (for 1 sec) - 1 sec OFF
Power ON
1. If you have completed all the installation tasks, switch on power supply.
• For PSUNI card installed in the system, connect the three-prong plug of the power cord from the
ETERNITY into the AC outlet, and switch on power supply.
• For PS48V card installed in the system, keep the MCB Switch ON and power the FCBC.
Reset Cycle
• Reset Cycle (Power-ON Self Test) takes about 2 minutes to finish.
• All the LEDs of the system, the cards and the keys of the DKP attached to the System are turned on.
Interpreting LEDs
The functioning of the LEDs of the system and the various cards and their meaning are summarized at the
end of the installation instructions for each Card Type.
Refer to the LED Patterns described for each Card Type to verify if the system is operating properly and locate
faults, where they occur.
When the reset cycle is successful, the default Station “Access Codes” loaded by the system and the date and
time of the “Real Time Clock (RTC)” of the system will appear on the LCD display of the DKPs you have
connected with the system.
• The Matrix ETERNITY is to be installed by persons who are trained and experienced in telecom wiring;
familiar with trunks, physical wiring of the MDF on both the PBX side and the line side (CO).
• Take all the necessary precautions for handling electronic and electrical appliances. Follow prescribed
procedures for preventing electrostatic discharges, to prevent damage to the cards and harm to
yourself.
• Wear an anti-static wrist strap/belt and use a grounding mat. Read the dos and don'ts listed in
“Protecting ETERNITY and Yourself”.
• If you have complied with the requirements and instructions described in “Before You Start”, you may
now begin the installation of your ETERNITY PE.
The Matrix ETERNITY PE is shipped factory fitted with the Power supply unit and the CPU Card (refer the section
“Know Your ETERNITY”).
The cards - SLT, DKP, TWT, BRI, T1E1PRI, GSM, VoIP, VMS, and the Door Phone Card - are shipped separately
as per the order placed by individual customers.
Illustrated below are the positions of the slots in each variant of ETERNITY PE.
ETERNITY PE6SP
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
Cards are mounted on the CPU and secured on the three studs on the CPU, with the screws provided.
The ETERNITY PE 6S has an Ethernet port, a Communication Port, an Analog Input Port and an analog Output
Port.
ETERNITY PE3SP
ETERNITY PE3SP is similar to PE6S, except it has only 3 universal slots.
ETERNITY PE3SS
ETERNITY PE3SS has 3 universal slots and an Ethernet Port.
Instructions are provided in the following for installation of the cards. These instructions are to be followed also
when you expand the system (add more cards) or remove cards for maintenance and repair.
1. Have all the necessary wiring ready. Read the topic Main Distribution Frame for guidance on how to set up
the MDF and connect the system with the MDF, and install Primary Protection against heavy voltages.
2. Unpack the box. Check the package contents (see Packaging List). Contact your Dealer/Distributor if any
of the items is missing, faulty or damaged. Do not discard the packaging material.
If you have decided to mount the ETERNITY PE on a wall, use the Mounting Template for drilling the holes
at appropriate distances on the wall.
4. Mount the system at the selected site. Make sure that the system is placed such that you have full access
to the front and back panels. The holes in the side panels are provided for ventilation; Make sure that these
are not blocked, to prevent overheating.
When installing the system in a rack49 allow adequate space between the system and other units for air
circulation.
6. Check the voltage at the power point from where the supply is to be given to the system. It should be as
per the specifications. Earth the system properly.
Inserting Cards
7. Make sure the power supply is turned off and the power cord is unplugged, before you begin inserting the
cards.
11. Grasp the card by its sides or corners. Fit the card into connectors of the selected slot. Ensure that the
card is seated perfectly for all the connector pins on the cards make complete contact with those on the
CPU (motherboard) on the bottom plane.
13. Following the above steps, install each card into the universal slots.
Detailed installing instructions are provided for each card - DKP, SLT, TWT, ISDN BRI, ISDN T1/E1/PRI,
GSM, VoIP, VMS - later in this section. Refer to them when installing each card type.
14. Using the cables supplied with the cards, connect the card interfaces with the MDF (for SLT, DKP and
TWT, Magneto), the NT1 termination device (for BRI lines), the ISDN modem (for T1/E1 PRI lines), the IP
network, a Computer as applicable for each card.
For cards with multiple RJ45 connectors, the cables are bunched together, but each cable is identified by a
distinct color marked at the Boot edge and the Insulation edge of the cable.
The color markings make it easy for you to identify the connector on the card into which the cable is
plugged into, and the hardware ports on that connector. For example, the cable marked with blue on the
Boot and Insulation edge is connected to the first port of the card, containing the hardware ports 01-04.
15. Lead the cables out of the enclosure through any of the two cable outlets on either side of the enclosure.
16. When you have completed inserting the cards and connecting the cables, replace the top cover by sliding
it in place. Secure the cover with the two screws you removed.
Since the connectors of the cards will not be visible after the cover has been replaced, you are advised to
label the cables appropriately to facilitate identification.
There is no provision for battery backup. So, you are recommended to provide Battery Backup of 24 VDC, 7-10A.H.
by connecting a UPS to keep the system powered during outages.
The Power Supply Unit is factory-fitted. It must be removed and refitted by trained technicians only, and
only for the purpose of fault repair or replacement.
The CPU
The CPU of ETERNITY PE manages the entire system, controls all other cards (SLT, DKP, TWT, DKP+SLT,
TWT+SLT, DKP+TWT, BRI, T1E1PRI Single, GSM, VoIP, etc.). All configuration and programming information is
stored on this card.
The CPU may be removed and reinstalled solely for the purpose of fault repair or replacement, and by
trained technicians only.
Connectors
The connectors of the universal slots of ETERNITY PE are located on the CPU. So, the cards of ETERNITY PE
must be mounted on the connectors on the CPU.
The CPU card has an Ethernet Port, a USB Port, a Communication Port, an Analog Input Port (AIP) and an Analog
Output Port (AOP), and a USB Port.
USB Port
The USB port functions as the Device port for downloading data on to the ETERNITY PE using a PC.
Communication Port
This is a single asynchronous, serial, full duplex RS232C communication port, labeled as COM 10101. The COM
Port has a DB-9 connector. The COM port is meant for connecting a PC to the ETERNITY PE to install and operate
Property Management Software (PMS), Call Accounting Software (CAS), and download Station Message
Recording (SMDR) reports and System reports, and Hotel Reports.
Ethernet Port (TCP/IP) RJ45 To connect to a PC or LAN (to run various software).
Communication (COM Port) DB-9 female To connect a PC (to run various software; download
system activity reports).
USB port (Device Port)a USB To download data on to the ETERNITY through a PC.
Analog Input Portb Standard audio jack To connect audio input device like an external music
source.
Analog Output Portc Standard audio jack To audio output device - Public Address/Paging System.
LED
The CPU has two dual color - Green and Red - LEDs labeled L1 and L2.
• L1 - indicates the health of the card during the normal functioning of the system.
• L2 - indicates the health of the card during the reset cycle. After power ON, when the system becomes
stable LED blinks Green for 1 second ON and OFF.
Jumper Number
Position Function
ETERNITY PE3SS ETERNITY PE3SP ETERNITY PE6SP
Do not change the position of Jumpers number J5 and J6 on ETERNITY PE3SS and PE3SP.
Do not change the position of the Jumpers J7 and J11 on ETERNITY PE6SP.
• access the web-based programming tool Jeeves from any PC on the LAN.
• set up and run software applications such as PMS and CAS on any PC on the LAN.
• generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
1. Connect the Ethernet Port of ETERNITY with the Ethernet Port of the stand-alone PC using the Ethernet
cable supplied with ETERNITY.
You need to connect to the PC via the Ethernet port for the following functions:
2. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
any standard Communication Cable.
You need to connect to the PC via the COM Port for the following functions:
When you connect the ETERNITY PE to a standalone/LAN PC, you need to make sure that
• The IP Address of the Master Ethernet Port of the ETERNITY PE and the Ethernet Port of the PC are
not the same.
• The Ethernet Port of ETERNITY PE and the Ethernet Port of the PC are in the same Subnet.
If the system is connected to a LAN PC, ask the LAN Administrator to assign an IP Address and a Subnet
Mask to the ETERNITY PE52.
• Change the IP Address and the Subnet Mask of the Ethernet Port by dialing the following commands from
a station of the ETERNITY PE.
• Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
You get programming tone.
To change IP Address:
• If there is a DHCP server on the LAN to which the Ethernet Port of the ETERNITY PE is connected,
there is no need to change the IP Address or Subnet Mask, as these will be provided automatically by
the DHCP server.
• You must only enable the DHCP flag of the Ethernet Port of ETERNITY PE.
• To enable DHCP flag
• Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
• You get programming tone.
• Dial 2117-1
• You get confirmation tone.
Connect a good quality external amplifier and matching speakers to the port. You may use a combination
of 10W amplifier and a 4W speaker.
Specification Value
Specification Value
• Plug in the audio jack of the device into the AIP connector.
Also refer the topics “Music on Hold (MOH)”, “Background Music (BGM)”, “External Music”.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages as a result of improper use.
The Single Line Telephone (SLT) Card provides the interface to connect as extension phones, any standard, two-
wire, analog single line telephone instrument-rotary, pulse-tone, cordless, feature phones with or without Calling
Line Identification.
The SLT Card is available in the following configurations for the models of ETERNITY PE. The SLT interface is
available in combination with Digital Key Phone ports and Two-wire Trunk ports on a single card.
ETERNITY PE Card Combination card, with 4 ports to connect to 4 Digital Key Phones and 4 Single
DKP4+SLT4 Line Telephones
ETERNITY PE Card Combination card, with 2 ports to connect to 2 Digital Key Phones and 6 Single
DKP2+SLT6 Line Telephones
ETERNITY PE Card Combination card, with 4 ports to connect to 4 Two-wire Analog trunk lines and 4
TWT4+SLT4 Single Line Telephones
ETERNITY PE Card Combination card, with 2 ports to connect to 2 Two-wire Analog trunk lines, 2 Digital
TWT2+DKP2+SLT4 Key Phones, and 4 Single Line Telephones
ETERNITY PE Card Combination card with 2 ports to connect 2 Two-wire Analog trunk lines, and 6
TWT2+SLT6 ports to connect 6 Single Line Telephones
Choose an SLT Card with the configuration that meets your requirement for SLT ports. Also consider the maximum
SLT Port capacity of the system you are installing.
The maximum number of SLT ports supported by the variants of ETERNITY PE are:
Connectors
The SLT Cards have RJ45 connectors. A multi-pair, cable is supplied for each connector on the card.
1. Decide the number of SLT extensions required and arrange for as many telephone instruments.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
3. Make sure that the power supply is switched off, before you begin the installation of the card. Always wear
an electrostatic discharge prevention wrist strap/belt and use a grounding mat.
4. Unscrew the top cover of the ETERNITY PE and slide it out. Keep the cover and the screws aside.
6. Grasping the card by its sides or corners fit it onto the connectors of the selected slot. The card should be
seated such that its connector pins make perfect contact with those on the CPU (motherboard) on the
bottom plane.
7. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
8. Repeat the same steps to install another SLT card. It is not necessary to install the other SLT cards in
subsequent slots. You may install the other SLT cards in any of the universal slots.
9. Now, use the cables supplied with the SLT card to connect the SLT wires with the Main Distribution Frame.
For each connector on the SLT Card, there is a separate cable with an RJ45 jack on one end and free at
the other end.
The color markings serve the purpose of identifying which cable is to be plugged into each connector, and
the wire-pairs for each port on the connector.
10. Plug in the RJ45 end of the SLT cables into the respective connectors with the help of the color markings
on the cables as illustrated for each SLT Card type.
11. Lead the cables out of the enclosure through any of the two cable outlets on either side of the enclosure.
Each wire-pair from the ETERNITY PE SLT Port must be terminated to the bottom of the Krone Connector,
while the wire-pair of the extension line to be connected to this port must be terminated on the top of the
Krone connector. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
13. If you have completed installing all cards, replace the top cover by sliding it in place. Secure the cover with
the two screws you removed.
• For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.
• When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the SLT will be
connected to the first port on the connector.
The Digital Key Phone (DKP) Card provides the interface to connect the proprietary digital key phones, EON, the
PC-based phone EONSOFT and the Direct Station Selection (DSS) Consoles with the ETERNITY PE.
The DKP Card is available in the following configurations for the models of ETERNITY PE.
ETERNITY PE Card Combination card, with 4 ports to connect to 4 Digital Key Phones and 4 Single
DKP4+SLT4 Line Telephones
ETERNITY PE Card Combination card, with 2 ports to connect to 2 Digital Key Phones and 6 Single
DKP2+SLT6 Line Telephones
ETERNITY PE Card Combination card, with 4 ports to connect to 4 Two-wire Analog trunk lines and 4
TWT4+DKP4 ports to connect 4 DKP/DSS Consoles
ETERNITY PE Card Combination card, with 2 ports to connect to 2 Two-wire Analog trunk lines, 2
TWT2+DKP2+SLT4 ports to connect 2 DKP/DSS Consoles, and 4 ports to connect 4 Single Line
Telephones
Select a DKP Card with the configuration that meets your requirement for DKP Ports. Also consider the maximum
DKP Port capacity of the system you are installing.
The maximum number of DKP ports supported by each variant of ETERNITY PE is:
1. Decide the number of DKP extensions and DSS consoles required and arrange for as many instruments.
2. Take all the necessary precautions for installing cards - wear an electrostatic discharge preventive wrist
strap and use a grounding mat. Make sure power supply is turned off and the power cord is unplugged.
4. Unscrew and remove the top cover of the ETERNITY PE, and keep it aside with the screws.
7. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
8. Repeat the same steps to install another DKP card. You may install the other DPK cards in any of the
slots, not necessarily in a sequence.
9. Now, use the cables supplied with the DKP card to connect the DKP wire-pairs with the Main Distribution
Frame.
For each connector on the DKP Card, there is a separate cable with an RJ45 jack on one end and free at
the other end.
The color markings serve the purpose of identifying which cable is to be plugged into each connector, and
the wire-pairs for each port on the connector.
10. Plug in the RJ45 end of the DKP cables into the respective connectors with the help of the color markings
on the cables as illustrated for each DKP Card type.
12. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY PE DKP Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the extension line to be connected to this port must be terminated on the
top of the Krone connector. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
13. If you have completed installing all cards, replace the top cover by sliding it in place. Secure the cover with
the two screws you removed.
Installing EON
Matrix offers EON, the proprietary digital key phone. EON is available in the following models:
• EON42
• EON48
EON42
• To mount EON42 on a wall, detach the Foot Stand on the bottom of the phone, by pressing the snap fits of
the foot stand backwards, and lifting it from its anchorage in the mounting holes.
• Now, insert the snap fits of the foot stand into the Wall Mount bracket slots on the bottom of the phone in
the “ wall up" direction.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON42. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
EON48
• To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
• When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
• Connect the handset of the EON48 to the phone body using the spring cord.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
3. Plug one end of the RJ45 cable supplied with the phone into the RJ45 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
4. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait'… appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
5. After successful handshaking and reset cycle, if the DKP Parameters have been programmed, the LCD
display of the EON will show the station number and the station name in a line. The day, date and time,
time zone in the other line.
202 Reception
M on 2 4 A U G 1 2 : 0 0
6. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, “Digital Key Phone-
Operation”.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
2. Place the DSS Console next to the DKP, EON, to which it is to be attached.
You can install two DSS consoles to a DKP. Refer “Direct Station Selection Console” for possible
combinations for installing the various models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated into the bottom of
the Krone Connector (of 'Station Lines' on the MDF).
5. The wire-pairs of the DSS Consoles should be terminated into the top of the Krone Connector (of 'Station
Lines' on the MDF).
Refer the topic “The Main Distribution Frame (MDF)” for illustration.
6. ETERNITY automatically assigns the first DSS Console discovered on the system to the first DKP, the
second DSS to the second DKP.
7. Only when two DSS Consoles are to be assigned to a single DKP, manual assignment of DSS to the DKP
is required. Refer “Configuring DKP Extensions”.
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
the speaker connectors into the dongle.
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
communication cable into the COM port of the computer.
5. Switch ON the computer. The computer must have Windows Operating System installed on it.
6. Now insert the EONSOFT CD-ROM supplied with this PC-based DKP into the CD drive of your Computer.
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
11. Click the shortcut to open the program. The EONSOFT window will open:
14. Select the COM Port to which the communication cable is connected.
This screen will appear only if the DKP port to which the EONSOFT is connected has been programmed
for parameters like Name, Station number, Date and Time.
• If this dialog box does not appear on the screen in response to the click the COM Port Option, test the
COM Port for data transfer.
• If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
• Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
• Click the button labeled 'Start Test' in the COM Port Settings dialog box.
• After clicking this button, observe the Test Result section in the dialog box.
The above screen shows that the COM Port/communication cable is working.
• If the 'Error Count' shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
• Remove the communication cable from the COM Port of the PC.
• Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
• Now, if the error count is zero, please check the Communication Cable.
• If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
The Two-Wire Trunk (TWT) Card provides the interface to connect the ETERNITY with the POTS Network. The
TWT Card supports the different standards and features of POTS Networks across the world.
The TWT Card is available in the following configurations for the variants of ETERNITY PE. TWT interface is also
available in combination with SLT ports on a single card.
ETERNITY PE Card TWT8 8-port card to connect 8 Two-wire analog Trunk lines from the CO network
ETERNITY PE Card Combination card, with 4 TWT ports to connect 4 Two-wire analog Trunk lines from
TWT4+SLT4 the CO network, and 4 SLT ports to connect 4 Single Line Telephones
ETERNITY PE Card Combination card, with 4 TWT ports to connect 4 Two-wire analog trunk lines and
TWT4+DKP4 4 DKP ports to connect 4 DKP/DSS Consoles
ETERNITY PE Card Combination card, with 2 ports to connect to 2 Two-wire Analog trunk lines, 2 ports
TWT2+DKP2+SLT4 to connect 2 DKP/DSS Consoles, and 4 ports to connect 4 Single Line Telephones
ETERNITY PE Card Combination card with 2 ports to connect 2 Two-wire Analog trunk lines, and 6
TWT2+SLT6 ports to connect 6 Single Line Telephones
Choose a TWT Card with the configuration that meets your requirement for TWT trunk ports, keeping in mind the
maximum TWT Trunk Port capacity of the system you are installing.
Connectors
The TWT Card has RJ45 connectors, with 4 TWT ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
1. Take all the necessary precautions prescribed for handling the cards and electronic equipment. Make sure
that power supply is turned off, and the power cord is unplugged before you begin the installation of the
card. Put on an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
3. Unscrew and remove the top cover of the ETERNITY PE, and keep it aside with the screws.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Repeat the same steps to install another TWT card. You may install the other TWT cards in any of the
universal slots, but not necessarily in a sequence.
8. Now, use the cables supplied with the TWT card to connect the TWT ports with the Main Distribution
Frame.
For each connector on the TWT Card, there is a separate cable with an RJ45 jack on one end and free at
the other end.
The color markings serve the purpose of identifying which cable is to be plugged into each connector, and
the wire-pairs for each port on the connector.
9. Plug in the RJ45 end of the TWT cables into the respective connectors with the help of the color markings
on the cables as illustrated below for each TWT Card type.
11. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Trunk Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY PE TWT Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the trunk line from the CO Network to be connected to this port must be
terminated on the top of the Krone connector.
Refer the topics “The Main Distribution Frame (MDF)” and “Terminating Trunk and Station Cables on the
MDF”.
12. If you have completed installing all cards, replace the top cover by sliding it in place. Secure the cover with
the two screws you removed.
The BRI card provides the interface to connect ETERNITY PE with ISDN BRI Network. The ISDN Network may be
a public or a private ISDN exchange.
The BRI Card is available in the following configuration for the variants of ETERNITY PE.
ETERNITY PE Card BRI2 2-Port card to connect 2 ISDN BRI Lines or ISDN Compatible Devices
The maximum number of BRI lines supported by each variant of ETERNITY PE are:
Connectors
The BRI card has 2 RJ45 Connectors. A separate cable is supplied for each connector.
1. Take all the necessary precautions prescribed for handling the cards and electronic equipment: turn off
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
3. Unscrew and remove the top cover of the ETERNITY PE, and keep it aside with the screws.
5. Seat the card onto the connectors of the selected slot. The connector pins of the card should make perfect
contact with those on the CPU (motherboard) on the bottom plane.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Repeat the same steps to install another BRI card. You may install the other BRI cards in any of the
universal slots, but not necessarily in a sequence.
ISDN
Network NT 1 BRI Port
ETERNITY
Power
U-Interface S/T
(2-wire) Interface
Customer Premises
Where,
• U Interface = between the NT1 equipment and the ISDN central office.
• S/T Interface = between the ISDN user equipment, in this case, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the ETERNITY.
TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY via
the NT1 device. Here, the ETERNITY is the Terminal Equipment, so the BRI Port must be programmed to work in
the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
• the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
• the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
BRI Line
NT BRI Port
ISDN (TE Mode)
Network
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be up to 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
ISDN NT
Network BRI Bus Bar
Terminal
BRI Port Resistance 100
(TE Mode)
Terminal
Resistance 100
ETERNITY ISDN Phone ISDN Phone ISDN Phone
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
• A maximum of 8 Terminal equipment or ISDN devices can be connected to a single NT on a bus up to 200
meters from the NT.
• 100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d < 1 Km
d1 < 30 meters
ISDN NT
Network BRI Bus Bar
Terminal
Terminal BRI Port Resistance 100
(TE Mode)
Resistance 100
Where,
TE = Terminal equipment of any ISDN Equipment
NT = Network Termination provided by Service Provider
TR = Terminal Resistance 100
d = distance from NT to the last TE Equipment
d1 = the total distance from first TE equipment and the last TE equipment.
In an Extended Passive Bus Configuration,
• You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
• However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
• Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
3. Use the cable supplied for each connector on the BRI card to connect the BRI Ports to the NT1 device
supplied by your ISDN service provider.
4. To connect the BRI Lines to the BRI ports, refer the configuration and pinout details given below for
guidance
4 Tx
5 Rx
3 Rx1
4 Tx1
5 Tx2
6 Rx2
Port 1:
1 NC Orange-White
2 NC Orange
3 Tx0A Green-White
4 Rx0A Blue
5 Rx0B Blue-White
6 Tx0B Green
7 VOut 0- Brown-White
8 VOut 0+ Brown
Port 2:
1 NC Orange-White
2 NC Orange
3 Tx1A Green-White
4 Rx1A Blue
5 Rx1B Blue-White
6 Tx1B Green
7 VOut 1- Brown-White
8 VOut 1+ Brown
Port 1:
1 NC Orange-White
2 NC Orange
3 Rx0A Green-White
4 Tx0A Blue
5 Tx0B Blue-White
6 Rx0B Green
7 VOut 0- Brown-White
8 VOut 0+ Brown
Port 2:
1 NC Orange-White
2 NC Orange
3 Rx1A Green-White
4 Tx1A Blue
5 Tx1B Blue-White
6 Rx1B Green
7 VOut 1- Brown-White
8 VOut 1+ Brown
The following diagram shows how to connect a BRI Line to the ETERNITY PE BRI port in the TE mode.
NT-1 ETERNITY
3
Rx1 TxA
4 4
Tx Tx1 RxA
ISDN
5
Network Rx Tx2 5 RxB
6
Rx2 TxB
NC V-
Power NC V+
• V- and V+ are used when a TE is connected to BRI port (in this case the port functions as network or
NT).
• For using the BRI port in the TE mode, the cable connections need not be crossed.
• When the BRI Port is to be used in the NT mode, the Tx and Rx cable of BRI port shall be crossed with
the Rx and Tx cables of the terminal respectively.
By default the Orientation Type of the BRI ports of ETERNITY are set as 'Terminals' (TE mode). So, you
may skip to the next step.
If the BRI Port is to be configured in the NT mode, all the related Jumpers should be set in BC position.
Refer the table below.
Jumper Position for BRI Port1 Jumper Position for BRI Port2
Mode
J1 J2 J4 J5 J7 J8 J10 J11
NT BC BC BC BC BC BC BC BC
TE AB AB AB AB AB AB AB AB
• When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
• When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
BRI Line NT
Network
• Last TE equipment
• Last point of the bus bar where the last TE equipment is connected.
• If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when:
• Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Jumper Position for BRI Port1 Jumper Position for BRI Port 2
Function
J3 J4 J3 J4
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position)
1 RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1 1 RJ45 Connector
ports on BRI Bus
Bar to which the
3 3
ISDN TE
4 4 Equipment is
connected
5 5
6 6
8 8
• The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
• Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
• Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
• Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
To do this, you must change the position of the Jumpers J1 and J2 on the BRI modules (daughter board),
of the BRI Card.
Jumper Position
Function
J1 J2
• The maximum power that can be fed to a single BRI port is 50mA.
• From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
• The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
10. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle.
The ETERNITY PE T1E1PRI Card provides the interface to connect ETERNITY PE to ISDN Network.
When connected to T1 carrier lines, the Card supports the following signaling types:
• PRI
• Robbed Bit Signaling
• Q-Signaling (QSIG)
• E&M
When connected to E1 carrier lines, the card supports the following signaling types:
• PRI
• Channel Associated Signaling (CAS)
• Q-Signaling (QSIG)
• E&M
The T1E1PRI Card is available in the following configuration for ETERNITY PE:
ETERNITY PE Card T1E1PRI 1-port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Single Compatible Device
The maximum number of ISDN PRI lines supported by the variants of ETERNITY PE are:
Connectors
The T1E1PRI card has an RJ45 Connector. A cable with RJ45 plugs on both ends is supplied with the card.
3. Unscrew and remove the top cover of the ETERNITY PE (if not opened already). Keep the cover and the
screws aside.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Repeat the same steps to install another T1E1PRI card. You may install the other T1E1PRI cards in any of
the universal slots, but not necessarily in a sequence. Any card can be inserted in any of the universal
slots.
Customer Premises
ISDN G.703
Modem
Network 4-wire HDSL DTE
4-wire PRI
(RJ-45 Connector) (RJ-45 Connector) Port
G.703
Modem
Power
• Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line, one at
the Local Exchange and other at the Customer's Premises.
• At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
• The DTE interface of the modem is to be connected to the PRI port (RJ45 connector on the Matrix
ETERNITY PE T1E1PRI Single Card).
9. Plug in one end of the RJ45 cable supplied with the card into the card's connector. Lead the cable out of
the enclosure through any of the two cable outlets on either side of the enclosure.
10. Plug the other end of the RJ45 cable into the Network Termination Unit.
11. Refer the following pin details for connecting the Network Termination Unit with the ETERNITY.
Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)
5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the installation guide of the HDSL Network Termination Unit being used by you.
The T1E1PRI Port of the ETERNITY PE terminates in an 8-pin RJ45, female connector and is wired
according to the figure below.
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
• To use the PRI Port for T1 connectivity, termination resistance must be changed to 100, by changing
the position of jumper J2 to AB position.
13. Repeat the same steps to connect another PRI card, if installed.
14. If you have no other card to install, replace the top cover, by sliding it in place. Secure the cover with the
two screws you removed.
15. If you have completed all other installation tasks. Power the system and observe the Reset Cycle.
The Mobile Card interfaces the ETERNITY with GSM/3G networks. It routes calls made and received over mobile
networks, like a mobile handset.
The card does not support GPRS features, Fax and Data services, network supported services, except
CLIR and USSD.
The Mobile card is available in the following configuration for ETERNITY PE:
ETERNITY PE Card GSM2 2-port card to connect to the GSM network (2 SIM Cards can be
installed)
ETERNITY PE Card GSM2 2-port card to connect to 3G network (2 SIM Cards can be installed)
3G
ETERNITY PE Card GSM4 4-port card to connect to the GSM network (4 SIM Cards can be
installed)
ETERNITY PE Card GSM4 4-port card to connect to 3G network (4 SIM Cards can be installed)
3G
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
The maximum number of Mobile ports trunks supported by each variant of ETERNITY PE are:
Antenna
For all four mobile ports, there is a single antenna with a male connector on the card. A splitter connects all the four
ports on the card into a single antenna. An antenna cable is provided.
• you are wearing an electrostatic discharge preventive wrist strap and have a grounding mat, before you
begin handling the card.
2. Get the SIM Card from the GSM service provider of your choice ready. Use SIM PIN protection, if required.
• change the SIM PIN to 1234 (this is the default PIN for all SIM cards used in the system). Changing the
SIM PIN to '1234' enables you to change the SIM PIN from the ETERNITY later.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Now, insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the
Mobile card. You can insert multiple SIM cards of the same GSM service provider or of different service
providers.
6. Remove the top cover of the ETERNITY PE, if not opened already. Keep the cover and the screws aside.
7. Select any of the free universal slots. Grasping the card by its sides or corners fit it onto the connectors of
the selected slot.
The card should be seated such that its connector pins make perfect contact with those on the CPU
(motherboard) on the bottom plane.
8. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
9. Repeat the same steps to install another Mobile card, in another free slot. It is not necessary to install the
Mobile cards in subsequent slots.
10. Connect the antenna cable provided with the card on the splitter connector on the card.
11. Lead the antenna cable out of the enclosure through any of the two cable outlets on either side of the
enclosure. Now, place the antenna at an appropriate location.
12. If you have completed installing all cards, replace the top cover by sliding it in place. Secure the cover with
the two screws you removed.
• At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
• Each time the Mobile Port sends a request, such as a Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the GSM network. It is fixed for 150 seconds for all Mobile ports.
The SIP-based VoIP Card enables the stations of ETERNITY PE to connect to the IP network and make Proxy as
well as Non-Proxy (Peer-to-Peer) VoIP calls. The card has a Registrar Server, allowing any SIP-based device to be
registered with it and function as an extension of the ETERNITY PE. With the VoIP Card, ETERNITY PE offers the
functionality of an IP-PBX.
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The VoIP card is available in the following configuration for the ETERNITY PE.
The LAN Port is used for connecting the VoIP Card to the Local Area Network to register SIP extensions through
the LAN Port.
The WAN Port is for connecting the VoIP Card to the public network over a Router/Modem. Any user on the public
network can be registered as SIP Extension through the WAN Port.
The LAN Port supports Static IP Addressing only. The WAN Port supports Static, DHCP and PPPoE IP
Addressing.
Voice Channels
There are 16 Voice Channels on the VoIP16 Card and 8 Voice Channels on the VoIP8 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
SIP Extensions
ETERNITY PE supports 50 SIP Extensions. Any SIP-enabled device like an IP-phone, a Softphone, analog phone
adapter, can be registered with the VoIP Card and function as the 'SIP Extension' of the ETERNITY PE.
The SIP Extensions function like extensions of the ETERNITY PE. SIP Extension users can make and receive calls
from and to other extensions of ETERNITYPE and external numbers over PSTN, GSM, VoIP and E&M lines53. You
can also connect the Standard and Extended IP Phones offered by Matrix as SIP Extensions.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic “License Management”.
A SIP Extension can be registered with the ETERNITY from three different locations. This helps organizations
overcome geographical distances and reduce call costs.
53. Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your coun-
try prohibit call traffic between the public telelphony networks and IP networks, you must configure Logical Partition in your system.
To know more, see “Logical Partition”.
• A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
If you want to make only Peer-to-Peer calls54 (calls made without the intervention of a SIP Server or Proxy
Server) you do not need the service of an ITSP.
If you have subscribed one or more SIP Accounts to make SIP calls, ask your Internet Telephony Service
Provider (ITSP) for the following information:
• SIP ID/User ID
• Authentication User ID
• Authentication Password
• SIP Registrar Server Address
• SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
• Network Information:
The card is located behind the NAT Router and Private IP is assigned to the WAN port.
54. Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
• IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
• IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
• Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
• Gateway Address
• DNS Address
• DNS Domain Name (if applicable)
VoIP Card connected to the Public Network for Matrix Extended IP Phones
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
Get these items/information ready before you install the VoIP card and connect it to the IP network.
1. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
3. Unscrew the top cover of the ETERNITY PE and slide it out. Keep the cover and the screws aside.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Using the Ethernet cable supplied with the VoIP card, connect the LAN and the WAN Port to the IP
network, which may be Public Internet or a LAN, or both.
• Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end into the Router/Modem.
• Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
• Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end of the cable into the Router/Modem.
• Connect the LAN Port of the VoIP Card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all installation tasks, replace the top cover by sliding it in place. Secure the cover
with the two screws you removed.
10. Switch on power supply. The ETERNITY PE VoIP Card does not have LED.
ETERNITY PE supports up to 50 SIP Extensions. The SIP Extensions function like DKP/SLT extensions of the
ETERNITY PE. SIP Extension users can make and receive calls to any extension user of the ETERNITY and to
external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP55.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY PE.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY PE and you must have the IP8
License. For more information on Licensing, see “License Management”.
You can register upto 50 SIP Extensions with a single VoIP Card of ETERNITY PE. However, at a time, only as
many extensions as the number of Voice Channels supported by the VoIP Card can make calls. For more
information, see “Voice Channels” under the description for the“The VoIP Card” for ETERNITY PE.
You can register the same SIP Extension from three different locations.
You may also connect the Standard and Extended IP Phones of Matrix.
The Matrix Extended IP Phone, SETU VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
• Background Music
• CO Call Waiting
• Hot Desking
• Live Call Screening
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
55. Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
“Logical Partition”.
• Connect SETU VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
• Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
• the Registrar Server Address of ETERNITY PE
• the Registrar Server Port
• the SIP ID
• Authentication ID and Password.
• Connect SETU VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
• You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
• Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
• use DHCP option 224 and Data Type as ‘String’ to provide Server Address to the Extended IP
Phones.
• Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY PE in the DHCP option.
• Log in to Jeeves. For instructions, read the topic “Using Jeeves” under Configuring ETERNITY.
• Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see “Configuring SIP Extensions”.
For instructions, see the topic “Matrix Extended IP Phone Settings” under Configuring SIP Extensions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SETU VP248. For the purpose of illustration, the premium model, SETU VP248P, has been used.
• When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
• Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
• Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
• When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
If you attach the Foot Stand at 45°, the phone will be placed in an almost upright position on your
desk.
• Decide which of these positions would work for you best and accordingly attach the Foot Stand.
• Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
• Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
OR
You may plug a headset with an RJ11 connector in to the headset port at the bottom of the phone.
5. Connect the LAN Port of SETU VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone56. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
When you power the phone, the boot process will be initiated in the following sequence.
• All keys with LED, including the Speaker key, and the Ringer LED, will glow.
Welcom e to M atrix
B ooting ...
• As soon as the ‘Loading...’ message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
• Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
• When the cursor is placed under the Extended IP Phone, press Enter key.
We l c o me to Ma t ri x
L oa din g V 0 5 R0 1 Ex t SI P
• After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See “Network Settings”at the end of
this topic.
• The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY PE.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
• On successful download of all configuration files, the phone attempts to register with ETERNITY PE.
• On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
1. During start-up, when the phone prompts you to change the network settings after loading the firmware.
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to ‘Local Menu’.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
CA03 * 0 #
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
• In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
• Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
• You can configure all network parameters described below, except the MAC Address.
Connection Type
• Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
• If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
To enter the dot/period in the IP Address, press the digit key ‘1’ twice.
Subnet Mask
• If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key ‘1’ twice.
Gateway Address
• If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
• If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
• If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
• If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key ‘1’ twice.
• If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
• If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
• This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
• If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
• ETERNITY PE CPU Card works as the Auto Configuration Server for the phone. Enter the IP Address or
the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as ‘String’ to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
• Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic57.
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
• Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
• To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
• Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
• Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
• Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
• Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
• Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
• Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
57. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), i.e. better handling
of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred to as
Class of Service (CoS) which is defined by IEEE 802.1P.
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see “Using PCAP Trace for Matrix Extended IP Phone”, under PCAP Trace.
When you change the Network Settings, the phone will restart.
• In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
N E T W O R K S TAT U S
MAC: 0 0:1 b:0 9:0 0:9 a:a 7
IP: 1 9 2 . 1 6 8 . 2 0 1 .2 0 5
MASK: 2 5 5 . 25 5 . 2 5 5 .0
G W: 1 9 2 . 16 8 . 2 0 1 .3
DNS:
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
• S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
PE.
• S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY PE.
The VMS Card provides a full fledged, 'in-skin' Voice Mail System with auto attendant and voice mail features. It is
designed to provide a variety of voice applications that are commonly supported by any external Voice Mail
System.
ETERNITY PE Card VMS16 Voice Mail System providing mailbox to all extensions of ETERNITY.
The card provides mailbox facility to all extensions of ETERNITY PE. Each Mailbox has the capacity of storing 254
messages. The size of each Mailbox is set by default to 5 minutes. The maximum message length for each mailbox
is set by default to 15 seconds58.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 1GB Pen Drive with the VMS
card. The Pen Drive supports 18 hours of recording.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or in a LAN) to access and use the
embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages. The Ethernet
Port can also be used for Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The USB port is used for connecting the
USB Pen Drive to the VMS Card. The pen drive is supplied by Matrix and contains all the voice messages, mailbox
messages, greetings and other messages and prompts.
The Pen Drive is factory fitted and shipped with the card.
2. Unpack the VMS card and check the package contents with the packing list.
3. Remove the top cover of the ETERNITY, if not opened already. Keep the cover and the screws aside.
4. Select from any of the free universal slots to install the VMS Card.
5. Hold the card by its sides and seat the card on the connectors of the slot on the CPU. Make sure that the
card's connector pins fit perfectly into those on the CPU.
58. When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of messages is 999 seconds.
Connecting to a Computer
7. Now, connect the card to a standalone PC/LAN.
• Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
• Lead the Ethernet cable out of the enclosure through any of the two cable outlets on either side of the
enclosure.
• Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a to a standalone/LAN PC, you need to make sure that
• The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the PC do not conflict,
are not the same.
• The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
The Door Phone Card is a special card for the ETERNITY PE models only. The card provides the interface to run
electronic gadgets (refer “Automated Control Applications”) and operate Door Phones with the ETERNITY PE.
The Door Phone Card supports 3 Door phone Ports, 3 Digital Output Ports, and a single Digital Input Port.
You can connect any standard 4-wire Door Phone with ETERNITY PE. Matrix does not supply Door Phones.
Specification Value
The DOP can act as a Relay for Door Locks. So, you can connect a Door Lock release device to the DOP to
work in conjunction with the Door Phone connected to the Door Phone Port.
Specification Value
The DIP can be used for certain “Automated Control Applications” and “Security Alarm and Reporting”. Read
these topics to know more.
Specification Value
2. Make sure that the power supply is switched off, before you begin the installation of the card. Always wear
an electrostatic discharge prevention wrist strap/belt and use a grounding mat.
3. Unscrew the top cover of the ETERNITY PE and slide it out. Keep the cover and the screws aside.
5. Grasping the card by its sides or corners fit it onto the connectors of the selected slot. The card should be
seated such that its connector pins make perfect contact with those on the CPU (motherboard) on the
bottom plane.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Now, plug in the cables supplied with the card into the connectors. Lead the cables out of the cable outlet
and terminate the free end of the cables into the Distribution Frame.
For each connector on the card, there is a separate cable with an RJ45 jack on one end and free at the
other end. Each cable is identified by a distinct color marked at the Boot edge and the Insulation edge of
the cable.
You can refer to the color markings to identify which cable is to be plugged into each connector, and the
wire-pairs for each port on the connector.
Refer to the pinout details of the ports on the Door Phone card to connect the wires appropriately.
1 Relay A Orange-White
DOP-1
2 Relay B Orange
3 Status Green-White
7 DIP_B Brown-White
DIP-1
8 DIP_A Brown
1 Relay A Orange-White
DOP-2
2 Relay B Orange
3 Status Green-White
1 Relay A Orange-White
DOP-3
2 Relay B Orange
3 Status Green-White
8. The wires of the devices - Door Phone, Door Lock, Sensor - you want to connect to the ETERNITY PE
should be terminated in to the Distribution Frame. Refer the pinout details of the ports of the Door Phone to
connect the cables.
• There is a door phone port on each connector of the Door Phone Card. You can connect any standard
4-wire door phone. Refer to the pin out details given above to make the connections.
• Make sure that the door phone you connect conforms with the Technical Specifications of the door
phone port.
• There is a Digital Output Port on each of the three connectors of the Door Phone Card. You can
connect a Door Lock or any other gadget you want to operate as an automated control application to
this port. Refer to the pin out details given above to make the connections.
• Make sure that the gadget you connect conforms with the Technical Specifications of the DOP.
• The Digital Input Port is located on the first connector of the Door Phone Card. Make sure that the
sensor of panic switch that you connect to the DIP conforms with the technical specifications of the DIP.
9. If you have completed all installation tasks, replace the top cover by sliding it in place. Secure the cover
with the two screws you removed.
Power ON
1. If you have completed all the installation tasks, connect the three-prong plug of the power cord from the
ETERNITY into the AC outlet, and switch on power supply.
Reset Cycle
• Reset Cycle (Power-ON Self Test) takes about 2 minutes to finish.
• The LEDs L1 and L2 are turned on, and glow Orange for a minute.
• The LEDs of the keys of the Digital Key Phones attached to the system are turned on sequentially.
When the Reset Cycle is successful, the default Station Access Codes loaded by the system and the date
and time of the Real Time Clock of the system will appear on the LCD display of the Digital Key Phones
have connected with the system.
• A telephone
Jeeves allows system configuration at three levels: System Engineer, System Administrator, and the Front Desk
User.
A distinct set of features and facilities can configured at each of these levels.
It is possible to configure the ETERNITY from any location using Jeeves. You can use Jeeves to configure
the system On-site (where it is installed) and Off-site, from a remote location.
To be able to do this, the System Engineer must enter the System Engineer (SE) configuration mode, by
logging into Jeeves as System Engineer.
Only the System Engineer, who installs, configures and maintains the PBX is allowed access to this mode.
Hence access to the SE mode is protected by means of a password, referred throughout this document as the
SE Password.
The SE password can be changed. Refer the topic “System Security” for instructions on how to change the SE
password.
If you forget the SE password, you can restore the default SE password. Read the topic “System Security” for
instructions on restoring the default SE password.
You are advised to record the SE password at a safe place, where it can be accessed by you (the System
Engineer) to avoid the inconvenience of restoring the default SE password.
• Quick Installation Wizard - Standard PBX: This wizard helps the Installer/System Engineer to quickly
set-up the ETERNITY for the standard PBX (Enterprise) Application.
Using this Wizard, the Installer/System Engineer can configure as much as 80 percent of the system
configuration, covering all the parameters necessary for the functioning of the system. For advanced
configuration of features and facilities, the Installer/System Engineer must use the “Using Full
Programming Access” mode.
Detailed information on this Wizard and instructions for using it for system configuration are given later in
this chapter. Refer the topic “Using Quick Installation Wizard - Standard PBX”.
• Hotel Installation Wizard - Hospitality PBX: This is a special wizard for the Hospitality Application of the
ETERNITY. The Hotel Installation Wizard simplifies the configuration process and helps the Installer/
System Engineer to configure the system for the special telephone and guest/patient management
facilities and features for hotels and hospitals.
You may use this Wizard if you have deployed the ETERNITY as a Hospitality Application. The Hotel
Installation Wizard is recommended to be used when configuring the ETERNITY for the first time.
The Full Programming Access mode of Jeeves, as the title itself suggests, enables the Installer to change the
values of all configurable parameters of the system, including those not covered by the Wizard.
Refer the topic “Using Full Programming Access” for more information and instructions.
For this, the System Administrator, who is usually the operator or receptionist, must login as System
Administrator into Jeeves.
SA Password
The access to SA mode may be protected by means of a password, referred to as the SA password in this
document.
The SA password is 4-digit code for preventing unauthorized access to the SA mode. The default, SA Password is
1111.
The SA password can be changed by the SA and the SE. However, if the SA forgets the password, the default
password or an entirely new password can be issued only by the SE.
Refer the topic “System Security” for instructions on how to change the SA password.
Four persons can simultaneously login and change system settings from the SA mode.
This mode is meant for the personnel at the Front Desk of the Hotels/Motels, allowing them access to and
operation of the hospitality features of ETERNITY, for example: Check-In/Out of guests, Changing Room
Occupancy and Clean Status, setting Call Budgets for guests, setting Wake-up calls, Reminders, Do Not
Disturb for guests, printing Call Reports and Hotel reports, and several others.
The system can be configured both, 'on site' and from a remote location (refer “Remote Programming”), and
from multiple extensions simultaneously, without its functioning being affected.
The ETERNITY can be configured by dialing the relevant command strings from a telephone. The telephone may
be any Single Line Telephone (SLT) or the Matrix proprietary Digital Key Phone (DKP), EON, connected as an
extension of the ETERNITY.
For the ease of operation, you may use EON instead of an SLT. Using EON has the advantage that
• you can view the command strings that you have keyed in on the LCD display of EON;
• you will get prompts and confirmatory messages on the phone's LCD display, in addition to confirmation
tone played to you.
System Configuration using a Telephone can be done at two levels: System Engineer Level and System
Administrator Level.
A distinct set of features and facilities can programmed at each of these levels.
It is possible to configure the ETERNITY from any location using a Telephone. You can use a Telephone
connected as an extension of the ETERNITY to configure the system On-site (where it is installed). You
can also configure the system using a Telephone Off-site, from a Remote location, using the “Direct Inward
System Access (DISA)” feature of the ETERNITY. You can access both the System Engineer mode as well
as the System Administrator mode from the remote location. Refer the topic 'Remote Configuration using a
Telephone' later in this chapter for instructions.
The System Engineer has Full Programming Access when configuring the system using a Telephone. In other
words, the System Engineer can configure the entire system, nearly all the programmable parameters, using
a telephone.
Refer the topic “Using Full Programming Access” for more information and instructions.
SE Commands
These are number strings to be dialed by the System Engineer on entering the SE mode via a telephone. SE
Commands are unique to the feature/facility being programmed. Hence SE Commands for configuring a particular
feature/facility are presented in description of that feature/facility in this manual.
Refer the topic “System Security” for instructions on how to change the SE password. In case the System Engineer
forgets the password, it can be restored to the default password and changed again. Read the section “Default
Settings” for instructions on restoring the default SE password.
There is no restriction on the number of persons who can simultaneously enter SE mode from a Telephone
and configure the system.
• If the SE Password you entered is incorrect, you will be played an Error Tone and an Error Message on
EON.
• The system accepts and executes the command immediately, but it takes approximately 2 minutes to
save a command. So, it is advisable that you do not turn OFF the system for 2 to 3 minutes after
entering the last command.
The Operator or Receptionist, who usually administer the system, must enter the System Administrator (SA)
mode by dialing 1#92-SA Password.
After entering SA mode, you may dial command strings referred to as SA Commands from a telephone.
SA Commands
SA commands consist of a prefix string 1072, followed by the Command string. For example: the SA command for
setting Do Not Disturb for an extension is 1072-001-extension number-1, where 1072 is the prefix string and 001
is the command string.
59. You may use the default SE password, 1234, if the password has not been changed already. If the password has been changed,
use the new password.
The command for entering SA mode (1#92) is also non-programmable. Refer “Access Codes” in the chapter
Features and Facilities.
To know how to use change feature settings with SA Commands, please refer the description of individual features
under “Features and Facilities”.
SA Password
The access to SA mode may be protected by means of an SA password.
The SA password is 4-digit code for preventing unauthorized access to the SA mode. The default, SA Password is
1111. It can be changed and reset by the System Engineer.
Refer the topic “System Security” for instructions on how to change and reset the SA Password using SE
commands.
• When the ETERNITY is used in the Standard PBX Application, you can enter SA mode only from
extensions which have the features 'SA mode' and or 'Allow SA Commands' enabled in their Class of
Service.
• When the feature 'Allow SA Commands' is enabled in the Class of Service of an extension, the
extension will always be in SA mode. You do not need to enter SA mode by dialing the SA password.
You can enter the SA mode by dialing the SA command prefix string.
• When the feature 'SA Mode' is enabled in the Class of Service of an extension, dialing of the SA
Password is required to enter the SA mode. SA Commands can be dialed only after successfully
entering the SA Mode.
• There is no restriction on the number of persons who can simultaneously enter and operate from the
SA mode using a telephone.
60. You may use the default SA password, 1111, if the password has not been changed already. If the password has been changed,
use the new password.
When you dial this command, the system will check if the facility 'SA Mode' is enabled in the Class of Service allowed to the exten-
sion from which you have dialed this command. If the SA mode is not allowed, an Error Tone will be played. The Error Tone will be
played also when the SA password is entered incorrectly.
If the facility 'Allow SA Commands' is enabled in the Class of Service of the extension you are using, you can skip this step and
directly dial the SA Command (1072-<Command String>).
You can exit from the SA mode automatically or manually. To exit the SA mode automatically, you must
program the SA Mode Timer. This Timer can be set to a desired duration between 000 and 255 minutes.
On the expiry of the set time, the system disconnects the extension phone from the SA mode. By default,
the SA Mode Timer is set to 003 minutes. The Timer is loaded every time a new SA command is issued.
• You can also exit the SA mode before the SA Mode Timer expires by dialing this command. If the SA Mode
Timer is set to 000 minutes, you can exit the SA mode only by dialing the command 1#92.
To do this:
• Dial 1#91-SE Password to enter SE mode from a DKP/SLT.
• Dial 2118-Time
Where,
Time is from 001 to 255 minutes.
For example, to set log out time to 45 minutes, dial 2118-045.
OR
• the Master Ethernet Port of ETERNITY must be connected with a stand-alone PC or in a LAN.
• a web-browser, either Internet Explorer 6 with SP2 (Service Pack) or Mozilla Firefox, must be installed on
the PC.
Refer the installation instructions for connecting a standalone/LAN as applicable to your model of ETERNITY
(“Installing ETERNITY ME”, “Installing ETERNITY GE”, and “Installing ETERNITY PE”).
When the system is connected to a standalone/LAN, change the IP Address and Subnet Mask so that the IP
Address of the PC and the Master Ethernet Port of ETERNITY do not clash and the PC and the Master
Ethernet Port of ETERNITY are in the same Subnet.
Refer the instructions provided under the sub-topic 'Changing IP Address and Subnet Mask of the Master
Ethernet Port' under the topics “Installing ETERNITY ME”, “Installing ETERNITY GE”, and “Installing
ETERNITY PE” as applicable to your model.
1. Open the browser (Internet Explorer/Mozilla Firefox) on the PC (Standalone or LAN PC) to which the
ETERNITY is connected.
2. Enter the current IP address of Master Ethernet Port of ETERNITY on the address bar or the browser.
4. You may select a language of your choice from the options provided on the top of this page. Jeeves
supports the following languages:
• English
• Italian
• Spanish
• French
• German
• Portuguese
English is the default language. When you select a language, all the pages of Jeeves will appear in the
selected language for the current session, until you log out of Jeeves.
You can also select a language on the Front Desk User Login page, the System Engineer Login page and
the System Administrator Login page. The language you select from these pages will also be valid only for
the current login session.
• The language that you select on the 'Welcome' page or on any of the Login pages is valid for the
session only. The default language will be applied on next login.
• When you select the “Region” for the country in which ETERNITY is being installed, the system will
load the country-specific default settings and automatically select the local language of the country, if
the local language is among the above listed languages. The default local language set on selecting
the Region will be applied for every login session, unless you select another language as the default
local language.
• The default local language set on selecting the Region can also be changed from the "System
Parameters" page of Jeeves.
Set the Appearance Settings of your PC's screen to the resolution of 1024 x 768 pixels for full view of the
pages of Jeeves, the horizontal and vertical scroll bars on the pages.
You may use the default SE password, 1234, if the password has not been changed already. If the
password has been changed, use the new password.
• Enter the desired programming option, by clicking the link. For instance, if you want to use the Quick
Installation Wizard-Standard PBX, click the link. The Wizard will open, you may navigate further.
• To exit the SE programming mode in Jeeves, click the 'Logout' button on bottom right of the browser
window.
• Each login session into the SE Mode is set to 60 minutes by default. So, the login session will expire at
the end of 60 minutes. The duration of the login session can be extended or shortened according to
your preference by changing the settings for the 'Web Configuration Timer'. Refer the topic “Changing
Login Session Time Out of Jeeves”.
• It is possible for four users to simultaneously log into the System Engineer Mode of Jeeves. It is also
possible to log out all of these users at once or log out any of these users selectively. Refer the topic
“Logging Out Users from Jeeves”.
• If you are using the model ETERNITY ME10S, with the Redundancy Option for the Master Card, you
will have two Master Cards; each card is loaded with the GUI Jeeves. Now, the Jeeves of the second
card can be accessed and used in the same way as described above. Make sure that the configuration/
settings of the active card are updated in the second card.
You may use the default SA password, 1111, if the password has not been changed already. If the
password has been changed, use the new password.
• On successful login, the Feature Menu for the SA mode is displayed as links on the left side panel as
navigation links on the page and on every SA programming page.
• Click the link of the desired Feature to open the page. Program the feature and click 'Submit' at the bottom
of the page to save changes. Click 'Default' to reset the default values.
• To exit the SA programming mode in Jeeves, click the 'Logout' button on the bottom right of the browser
window.
File Transfer Protocol (FTP), is a standard network protocol, used to exchange and manipulate files over a TCP
computer network such as the Internet.
FTP is the simplest way to exchange files between computers on the Internet. Like the Hypertext Transfer Protocol
(HTTP), which transfers displayable Web pages and related files, and the Simple Mail Transfer Protocol (SMTP),
which transfers e-mail, FTP uses the Internet's TCP/IP protocols, and is commonly used to transfer Web server for
everyone on the Internet, download program and other files to your computer from other servers.
Using FTP, you can also update (delete, rename, move and copy) files at a server. For this, you need to log on to
an FTP server.
ETERNITY supports an embedded FTP server which can be used for three purposes:
Configuration of ETERNITY using FTP is meant for Installers who want to complete system configuration at their
end or are unable to configure the system on-site at the customer's end.
The advantage of using FTP for configuring the system is that Installers can complete the entire system
configuration as per their customer's requirement at their end, copy these configuration files, and then upload these
configuration files on to the customer's system.
Further, once the configuration file has been created, Installers only need to make the desired changes in the
relevant files and upload the updated files again.
Before you configure the system using the FTP server, make sure you have completed the following tasks at your
end:
• Enter the IP Address of the Master Ethernet Port of ETERNITY in the Address Bar to open Jeeves. Refer
“Using Jeeves” for instructions.
• Configure the system as per the customer/user requirements “Using Quick Installation Wizard - Standard
PBX” or “Using Full Programming Access”.
• The configuration files have the extension '.cfg'. Select all files in the folder. Copy the files.
OR
• Create a new folder in the selected location and rename the folder.
• Copy the configuration folder you created on a CD or a Pen Drive for the purpose of uploading the
configuration files on to the customer's system.
If you are an Installer who has multiple customers and you want configure your customers' systems using
FTP, you are recommended to tag the names of the configuration folders you create for your customers
with some identification, like name and date.
• Enter the IP Address of the Master Ethernet Port of ETERNITY in the Address Bar to open Jeeves. Refer
“Using Jeeves” for instructions.
• On the left side panel containing the SE mode menu options, scroll down to reach 'Configuration Upload'.
• On successful login the FTP window will open. The all the system configuration files are stored in the
folder named 'config'. The configuration files have the extension '.cfg'.
• Select all files/files in the folder on the FTP and delete the files.
• Now, copy the files you created for the customer from the location it is stored (from the CD-ROM drive/
folder, if the files are on a CD; Disk/folder, if saved on a disk drive of the PC).
• Go back to the 'config' folder, and paste all the files you copied on to this folder.
The FTP can be used to store back-up of configuration files, SMDR, and System Software files. Refer the
topics “Backup-System Configuration”, “Backup-SMDR” and “Backup-System Software” to know more.
The ETERNITY can be programmed via the Serial COM port of a computer, using communication software like
HyperTerminal, ProComm, and BitComm.
For this, the ETERNITY must be connected to the Communication (COM) Port of a PC.
You must create a Text file and enter the programming command strings in the file. The Text file must be uploaded
on to the ETERNITY using the communication software, such as HyperTerminal/ ProComm/ BitComm.
The advantage of programming the ETERNITY using a computer is that once the Text file has been created, you
only need to make any subsequent changes that are required at the relevant places in the file and upload it again
on the PBX.
To program the ETERNITY from the Serial Port, follow these steps:
1. Connect the communication port of the ETERNITY with the communication port of the computer (any
standard computer with Window and NT operating system) using a crossed communication cable with 9-
pin D-type female connector on both ends.
2. Enable 'Programming through COM Port' in the ETERNITY for serial communication, select the COM Port
for programming, and program the COM Port Attributes of ETERNITY and that of the PC. You can do this
in two ways:
a. Using Jeeves
• In the 'Communication Port for Programming' box, select the port you want to use for programming:
COM Port 1, COM Port 2. By default no COM Port is selected.
b. Using SE Commands
• Open Notepad (point cursor on Start Programs Accessories; click NotePad, a new, untitled file will
open).61
• Each command string should be prefixed with the character '^' (press shift+6)
• After you have entered the SE commands, save the text file containing the commands with an
appropriate file name with .txt extension. Name the file such that you will know what commands it
contains. For example: If you have entered SE commands to configure SLT ports, save the file as SLT-
port-parameters.txt.
5. Enter the name you desire to give to the new connection and select an icon. Click OK.
If you have ignored the location information in Step 3, you may be prompted again. You may ignore the
prompt.
7. Configure the 'Port Settings' - Bits per second, Parity, Data Bits, Stop Bits, Flow Control. Remember, these
values must match with the values you entered for the COM Port of the ETERNITY to which this COM Port
is connected. Click 'Apply' and click 'OK'.
11. Click 'Transfer' on the task bar of HyperTerminal window. Select the option 'Send Text File' from the drop
down menu. The 'Send Text File' window will open.
13. The computer is now sends the command file character by character. The ETERNITY receives these
characters and executes commands after assembling them in a command format.
If the PBX accepts the programming, it responds by sending the character 'Y'. If the PBX does not accept
it, it responds by sending character 'N'.
16. You may verify the programming of ETERNITY by logging into Jeeves and checking the parameters of the
feature/facility you programmed using HyperTerminal. For example, click SLT Parameters page and check
if the command strings you uploaded in the text file SLT-port-parameters.txt are reflected here.
The Quick Installation Wizard-Standard PBX helps you with the basic configuration of the system in easy steps. It is
designed to break down the complexities of programming and can cover as much as 80% of your basic installation
requirements.
The advantage of using the Wizard is that it speeds up system configuration, as you configure parameters that are
specific to your model of ETERNITY and only those parameters you would like to use. For example, if you are
installing ETERNITY PE3SS, the Wizard will not display ISDN BRI and T1E1 PRI parameters, as these are not
supported in this model. Similarly, the Wizard will show only the Trunks and Extensions that are connected to the
system for configuration, for instance, if you are installing only 12 out of the 128 TWT trunks supported by
ETERNITY ME16S, and 12 DKP and 120 SLT extensions, the Wizard will show only as many trunk and extension
ports for configuration. Thus, the Wizard makes the entire system configuration very focused.
Further, the Wizard can be used for system configuration for first time installation as well as any time post-
installation to make modifications in the system configuration.
The Quick Installation Wizard - Standard PBX offers the configuration of the following parameters:
1. Region
2. System Pre-requisites
3. Extension Numbering Plan
4. Naming Trunks
5. Day-Night time
6. Number Patterns
7. Operator
8. Extensions
9. Door Phones
10. Least Cost Routing (LCR)
11. CPU Groups
12. TWT Trunks
13. BRI Trunks
14. T1E1PRI Trunks
15. Mobile Trunks
16. VoIP Network
17. SIP Trunks
18. Emergency Numbers
Help Text
There is a help text to explain the parameters on each screen. You may use this help text to guide you in entering
the information.
• Next: Clicking this button will cause the existing values of the parameters in a page to be submitted and
takes you to the next page of the Wizard.
• Submit: Clicking this button will cause the parameters configured in the page to be submitted, but will not
take you to the next page. The 'Submit' button is to be used when there are multiple pages for configuring
a facility, for example, configuring extensions, TWT trunks. Instead of navigating all the pages, you can
access the desired page, make changes and submit them. Similarly, if you want to make any changes
post-installation, instead of navigating through all pages of the Wizard and clicking 'Done' to effect the
changes, you can reach the desired page, make the necessary changes and submit the changes.
• Skip: Clicking this button will take you to the next page, without making any changes to the current page.
The Skip button is to be used when the Wizard is used post-installation to reach a page which is to be
modified, without changing the existing configuration settings on other pages.
• Undo: Clicking this button refreshes the page with the parameters configured in the system. You may use
this button if you are not sure about the values you entered or have entered incorrect values and wish to
start all over again. This button is available only on select pages of the Wizard.
• Help: Clicking this button on a page opens a dialog box, containing Help Text for that page. The dialog box
can be maximized.
• Default: Clicking this button will populate all the fields of the page with the default values. Use this button
when you want to assign default values to parameters.
• Clear: Clicking this button will cause the values of all the fields on the page to be cleared. Use this button
to make corrections or start entering the values all over again.
• Cancel: This button appears on the dialog box that pops up on a page. Clicking this button on the dialog
box will close the box, while keeping you on the same page.
• Exit: Clicking this button will cause you to exit the Wizard.
The changes you make in the system configuration “Using Full Programming Access” will not be reflected
in the Quick Installation Wizard.
You are advised to either use the Wizard only during installation and to make modifications post
installation OR use the Wizard only for basic configuration (first time installation) and the Full Programming
Access mode for making subsequent changes.
• Click the Quick Installation Wizard-Standard PBX to open the Wizard. The Welcome page of the Wizard
will open.
Configuring Parameters
The Wizard gives you two options for configuring the system:
• Selective Configuration - This option allows you to choose what you want to configure and click the
related links on the navigation bar of the Wizard page. You are recommended use this option for making
the desired changes post-installation.
• Navigation by Wizard - In this option, the Wizard will lead you logically, step-by-step through the
configuration of the desired parameters. You are recommended to use this option when installing the
system for the first time, using the Wizard.
When you are installing the system for the first time, follow the steps given below to configure the system.
Region
• In the 'Region' list, select the country where the system is installed, and then click 'Next'.
The system will restart and load the default values according to the 'Region' you selected on the previous
screen.
If you are using the Wizard post-installation, do not default the system. This will roll back the entire system
configuration to the default values. Use the 'Skip' button, to move to the next page, without effecting
changes on the parameters of the current page.
• Enter the “Customer Name”. You may enter the name (and address, if desired) of the organization/
enterprise in this field. For example: Prudent Investment, 701 Sunshine Boulevard, Bannerghatta,
Bangalore. The Customer Name can be a maximum of 80 characters.
• In 'Model Type', select the model of ETERNITY you are configuring. For example, ETERNITY ME16S.
When you select the 'Model Type', the Wizard will display the different port types according to the system
capacity (the maximum number of ports) supported by the model and variant of ETERNITY you have
selected as the Model Type. For example, if you selected ETERNITY GE6S as Model Type, the maximum
number of TWT Ports will be 96 and the maximum number of DKP extensions would be 96 ports. If you
selected ETERNITY PE6SP, the maximum number of TWT Ports will be 16, and the maximum number of
DKP ports will be 48.
• Select the number of ports to be used for each Port Type (TWT, DKP, SLT, Mobile, T1E1 PRI, BRI, VoIP,
SIP Trunks, SIP Extensions) from the respective boxes. For example, if you want 8 TWT Trunks, 24 DKP
extensions, and 128 SLT extensions to be used, select the same numbers from the box.
If you want to use voice mail, or connect any device to the Digital Input Port or the Digital Output port,
select 'Yes'.
• If you have selected the System Pre-requisites, navigate to the next page of the Wizard by clicking the
'Next' button.
• When the 'On-Site Configuration?' flag is enabled, the Wizard will display the only those ports that are
present in the system (detected by the system at Power-ON) on this page.
For example, the system detects ETERNITY ME SLT32 card at power-on, so the maximum number of
SLT ports in the box will be 32. If you want only 20 SLTs to be used, select 20 in the box.
The Wizard will now consider that there are only 20 SLTs in the system and will modify the relevant
pages accordingly.
To cite another example, if your system is ETERNITY PE3SS, which does not support ISDN BRI or
PRI, the fields of these port types and ISDN Terminals on this page are displayed as non-editable.
Similarly, the field of the Door Phone ports will be editable only if the system has detected a Door
Phone card.
The fields for port types which are not available on-board (detected at Power-On) are displayed as non-
editable.
• To enable the 'On site Configuration' flag you must enter Full Programming Access mode.
• The Wizard does not provide for the port types 'Magneto' and 'E&M' as these are less commonly used.
If your system has Magneto card and or E&M card installed, you must configure the related trunk
parameters from the Full Programming Access mode only.
It is recommended that you enable the 'On-Site Configuration?' flag when you are configuring the system
at the installation site.
The desired Extension Number can be 6 digits long. The digits 0 to 9, # and * are allowed.
The desired Extension Name may be the name of the person who will use the extension. The name can be
a maximum of 18 characters.
When you change the extension numbers, make sure that they do not clash with any other Feature Access
Code in the dialing phase. To know more, refer the topic “Access Codes”.
The Wizard will display only those ports available in the system and the number of ports you have defined
earlier in the System Pre-requisites page. The Wizard automatically detects the Hardware Slot and Port
Offset of the ports and assigns them to Software Ports. The Wizard also assigns the default extension
numbers to the ports, but leaves the extension names blank.
The default extension numbers for the above port types are:
2601 to 2603 for Door Phone 1 to 3.
2001 to 2512 for SLTs 001 to 512.
3001 to 3128 for DKPs 001 to 128.
3201 to 3264 for ISDN terminals 01 to 64.
3301 to 4300 for SIP Extensions 001 to 999.
In the case of SIP Extensions, the number you program as Extension # will be considered as the SIP ID,
the Authentication ID and the Authentication Password.
If more than one VoIP Card is present in the system, by default, all SIP Extensions will be assigned to the
first VoIP Card that is detected by the system. The System Engineer may change this assignment, if
required.
• To assign the extension numbers and names all over again, click the 'Clear' Button on this page.
Access Codes
• Change the Feature Access Codes, if required. Feature Access Codes may consist of single digits or a
sequence of a maximum of 6 digits. Digits 0 to 9,# and * are allowed. The default Access Codes that
appear on the screen are country-dependant. Also, refer “Access Codes” to know more.
If you assign the same Access Code to more than one feature, the Wizard will pop up a “Total Conflict’
message and ask you to resolve the conflicting codes. It will not allow you to submit until you have
resolved the conflict. If you assign an Access Code which has the same first digit as another Access Code,
the Wizard will pop up an alert about the clashing numbers. You may choose to resolve the clashing
number by clicking ‘Cancel’ or you may ignore the alert and continue programming by clicking ‘OK’.
The Wizard automatically detects the Hardware Slot and Port Offset of the trunk ports and assigns them to
Software Ports. The Wizard also assigns the default trunk port names along with their respective software
port numbers. For example, if it is a Two-wire Trunk assigned software port number 001; the name will be
displayed as TWT -001. Thus TWT trunks are named as TWT-001 to TWT-128, Mobile Ports as MOB-001
to 0064, BRI Ports as BRI-001 to BRI-032, and so on.
The Wizard will display only those Trunk port types available in the system and the number of Trunk ports
you have defined for each trunk port type earlier on the System Pre-requisites page.
Day-Night Time
• Ask your customer about their working days and working hours (24 hours format) and program the
parameters accordingly. The Wizard considers the working hours you have selected as Daytime and the
• Day-Night Time is assigned to Trunks and Extensions, so that they behave differently according to the
Time of the day. For example, the customer may want the system to route trunk calls to the security
personnel when the office is closed, or deny certain extensions access to outgoing long distance calls
during non-working hours and days, or to play a different greeting message to callers on holidays.
• This parameter is based on Time Table 162, which is assigned to Trunks and Extensions by default.
• The Wizard simplifies the assignment of Time Tables to trunks and extensions, requiring you to
program only the working hours and working days, instead of prompting you to define the non-working
hours and break hours. The Wizard automatically applies the working hours and days you have
programmed to time table-dependent facilities and features such as Class of Service, Toll Control,
Outgoing Trunk Access, etc.
• Skip this page if you feel that the requirements of your customer are not served by this parameter63.
Configure the Time Tables and related features like Class of Service, Toll Control, Outgoing Trunk
Access, etc. from the Full Programming Access mode.
• If you want to change the working hours and days at a later stage while navigating the Wizard, you may
use the 'Back' button of your browser to return to this page and make the changes.
62. A Time Table is a schedule of the three time zones (working hours, break hours, non-working hours) for a week. There are 8 differ-
ent Time Table templates to select from. Different Time Tables can be assigned to different trunks and extensions. Refer the sec-
tion “Time Tables” to know more.
63. For instance, the working hours are not the same throughout the week.
Number Patterns
This parameter is related to the “Toll Control” Feature, which allows you to define a particular calling permission for
each extension, referred to as 'Call Privilege'. A Call Privilege allows the extension to call certain areas and restricts
it from calling others. The extension can also be restricted from the dialing of specific telephone numbers. The
ETERNITY supports six types of Call Privileges, these are: No Calls, Local Calls, Regional Calls, National Calls,
International Calls and Limited Calls.
• On this page, you are required to define the number strings which the system should consider as Local
Numbers, Regional Numbers, National Numbers and International Numbers.
In the field 'Numbers starting with', you may enter only the first digit of the number string, or a part of the
string, or the complete number string.
In the field 'except', enter the number strings which you want to restrict from being dialed out.
Each number string you enter must not exceed 16 characters. Separate number strings with comma.
• The Call Privilege Type 'No Calls' does not require any number pattern programming.
'Operator'
• Select the extensions which are to be used as Operator.
When you double click this field, a list box will open. The box on the left displays the extension names and
numbers you programmed or the default access codes. Place your cursor on the desired extension and
click the 'Select>>' button. The selected extensions will appear on the box on the right.
You will be shown an alert if you program more than 32 extensions. You may select and delete the excess
extensions or choose to ignore the alert by clicking 'OK' or closing the alert dialog box. Regardless of this,
the system will consider only 32 extensions.
When you click 'OK', all the extensions you selected will appear sequentially, separated by comma in the
order you selected the extensions. You can change the sequence of the extensions on the right hand side
using 'drag and drop'.
• Define the Class of Service for the Operator extension. Select the features to be allowed to the Operator
extension during the day and at night by selecting the checkboxes of the features listed in the table.
• Configure the Toll Control for the Operator extension for day time and night time. Select the type of calls to
be allowed during the day, and the type of calls to be allowed during the night. By default, all types of calls
are allowed during the day and at night.
• Select the Outgoing Trunks for the Day Time (the trunks through which calls are to be routed during the
day). If you want to the system to use Least Cost Routing, click the check box.
As you can see, the same trunk types are arranged sequentially, regardless of their hardware port location.
If you select trunks of the same type in sequential order, for example: TWT001, TWT002, TWT003, BRI01,
BRI02 and MOB02, the same trunk type will be grouped in one OG Trunk Bundle: TWT001, TWT002,
TWT003 will be OGTB #1, BRI01 and BRI02 will be grouped as OGTB#2, and MOB02 as OGTB#3.
These are default trunk names. If you have changed the trunk names (see Naming of Trunks), the name
will appear here, instead of the default trunk names.
If you select the trunks of the same type in a non-sequential order, for example, TWT001,
MOB01, BRI01 and TWT002, four OGTBs will be formed with TWT001 as member of OGTB#1, MOB01 as
member of OGTB#2, BRI01 in OGTB#3 and TWT002 in OGTB#4. So, despite two same trunk types being
selected (TWT001 and TWT002) they are grouped in separate OGTBs.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
It is possible to change the sequence of trunks on the right side box using the drag and drop action. You
can also delete a particular trunk from the right box.
Follow the same instructions for selecting the trunks from the list box as described in the previous step.
• Set the “Priority”64 for the Operator extension, by selecting the desired Priority Level from 1-9 from the box.
By default the priority level for the Operator extension is set to level 5.
64. Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3, 4...to 9. With 1 being the lowest priority and 9
being the highest priority. The calls from an extension with higher priority has preference in call landing. When an extension with
higher priority calls another with lower priority, a triple ring is placed on the called extension, and the call will land first on the exten-
sion when there are multiple incoming calls on the extension with lower priority.
For the parameters of the Operator extension, the Wizard uses the following resources:
• Class of Service Groups 18 to 19.
• Outgoing Trunk Bundle Groups 18 and 19
• Outgoing Trunk Bundles 69 to 76.
When programming the system from the Full Programming Access mode, do not modify the settings of
these parameters. It will affect the settings made by the Wizard.
Extensions
• Create a User Profile for extensions. A User Profile consists of Class of Service (COS), Toll Control and
Trunk Access to be assigned to an extension during day time and night time.
The Wizard makes configuration of the extensions easy with 'User Profiles'. Instead of configuring each
extension individually, you can group together extensions that are to be allowed the same COS group, Toll
Control and Trunk Access (Outgoing Trunk Bundle and Outgoing Trunk Bundle Group) in a single User
Profile.
User Profiles meet the requirements of organizations that desire to assign a different set of features to their
personnel according to their position in the organization, like senior managers, field executives,
administrative assistants etc. In such cases, each of these groups of users can be assigned a different
User Profile.
You can name each User Profile such that it reflects the extension user group to which it is assigned. For
example, you may rename User Profile-1 created for managers as 'Manager', User Profile-2 created for
field executives as 'Executive', User Profile-3 created for administrative assistants as 'Admin'.
• To change the label of the User Profile, click the 'Pencil' icon.
• A prompt will pop up, asking you if you want to rename the User Profile Name. Enter the desired name in
the field. You can enter a maximum of 18 characters as name. Click 'OK'.
• The name you entered will appear in place of 'User Profile' number.
• To assign the User Profile to the desired extensions, double click the box.
The Wizard will display the configured extension numbers on the left box. Place your cursor on the desired
extension numbers and click 'Select'. The selected extension numbers will appear on the right box.
You can select the extensions for the User Profile after configuring the other parameters of the User
Profile.
• Define the Class of Service for the User Profile for day time and night time by selecting the check boxes of
the features you want to allow in the Class of Service.
• Define the Toll Control for the User Profile for the day time and night time by selecting the desired Toll
Control level in the box.
• Select the Outgoing Trunks for the Day Time (the trunks through which calls are to be routed during the
day). If you want to the system to use Least Cost Routing, click the check box.
When you double click the field, the list box will open. Select the outgoing trunks from the left box.
As you can see, the same trunk types are arranged sequentially, regardless of their hardware port location.
If you select trunks of the same type in sequential order, for example, TWT001, TWT002, TWT003, BRI01,
BRI02 and MOB02, the same trunk type will be grouped in one OG Trunk Bundle: TWT001, TWT002,
TWT003 will be OGTB #1, BRI01 and BRI02 will be grouped as OGTB#2, and MOB02 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, such as: TWT001, MOB01, BRI01 and
TWT002, four OGTBs will be formed with TWT001 as member of OGTB#1, MOB01 as member of
OGTB#2, BRI01 in OGTB#3 and TWT002 in OGTB#4. So, despite two same trunk types being selected
(TWT001 and TWT002) they are grouped in separate OGTBs.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
It is possible to change the sequence of trunks on the right side box using the drag and drop action. You
can also delete a particular trunk from the right box.
• Select the Outgoing Trunks for the Night Time (the trunks through which calls are to be routed during the
night). If you want to the system to use Least Cost Routing, select the check box.
Follow the same instructions for selecting the trunks from the box as described in the previous step.
• Set the “Priority”65 for the extension, by selecting the desired Priority Level from 1-9 in the box. By default
the priority level for the extension is set to level 5.
You may set a different Priority Level in each User Profile, depending on the requirement of the extension
users, whose extension the User Profile is to be assigned. For example, you may set a higher Priority level
to the User Profile to be assigned to Managers.
• Rename the label, if required, by clicking the 'Pencil' icon. Repeat all the steps described above to define
Class of Service, Toll Control, Trunk Access and Priority.
• Click 'Submit' to save changes. Repeat the same steps to program another User Profile.
• When you have finished programming the desired number of User Profiles, click 'Next' button to navigate
the Wizard further.
65. Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3, 4...to 9. With 1 being the lowest priority and 9
being the highest priority. The calls from an extension with higher priority has preference in call landing. When an extension with
higher priority calls another with lower priority, a triple ring is placed on the called extension, and the call will land first on the exten-
sion when there are multiple incoming calls on the extension with lower priority. Refer the feature description for “Priority” to know
more.
• The User Profiles 1-8 use the following resources in the system:
• Station Basic Feature Template number 2 to 9
• Class of Service Groups 2 to 17.
• Outgoing Trunk Bundle Groups 2 to 17
• Outgoing Trunk Bundles 5 to 68.
• Do not use them when configuring the system from the Full Programming Access mode.
Door Phones
• Configure the following door phone parameters for each Door Phone you have connected to the Door
Phone Port of the ETERNITY PE.
• Route Door Phone Calls: Select the mode of routing for calls landing on the Door Phone.
Select 'At Wish' if you want the flexibility to have calls routed to a group of extensions or to an external
number as you desire.
Select 'As per Schedule' if you want the system to route the Door Phone Calls automatically to landing
destination phone (extension or external number) according to the time of the day.
• Route Door Phone Calls during Day to: Select the radio button of the desired destination - External
Number or Extensions - on which calls should be landed during the day time.
• Route Door Phone Calls during Night to: Select the radio button of the desired landing destination -
External Number or Extensions - for calls during the night time.
• Extensions for Day Time: Select the Extension numbers where calls should land during the day.
When you double click this field, the multiple selection box will open. The Wizard will display the
extensions you have configured on the left box. Place your cursor on the desired extension and click
select. The selected extensions will appear on the right box. You can select a maximum of 32
extensions as the landing destination for the door phone calls.
• Extensions for Night Time: Select the Extension numbers where calls should land during the night.
To select the extensions, double click the field, the multiple selection box will open. Place your cursor
on the desired extensions appearing on the left side box and click ‘select’. You can select a maximum
of 32 extensions. The selected extensions will appear on the right side box.
• External Number: Enter the External Number. This number is common for Day time and Night Time,
which means, you cannot have calls routed to different external numbers for the Day Time and Night
Time.
When you click the field 'To Dial External Number, Use Trunk:', the box will open, displaying the trunk
types present in the system in the left side box. Click the desired trunk type and click ‘Select’.
If you select trunks of the same type in sequential order, for example: TWT001, TWT002, TWT003, BRI01,
BRI02 and MOB02, the same trunk type will be grouped in one OG Trunk Bundle, for example: TWT001,
TWT002, TWT003 will be OGTB #1, BRI01 and BRI02 will be grouped as OGTB#2, and MOB02 as
OGTB#3.
If you select the trunks of the same type in a non-sequential order, for example: TWT001,
MOB01, BRI01 and TWT002, four OGTBs will be formed with TWT001 as member of OGTB#1, MOB01 as
member of OGTB#2, BRI01 in OGTB#3 and TWT002 in OGTB#4. So, despite two same trunk types being
selected (TWT001 and TWT002) they are grouped in separate OGTBs.
It is possible to change the sequence of trunks on the right side box using the drag and drop action. You
can also delete a particular trunk from the right box.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
• Select the check box of 'Use LCR?' to enable Least Cost Routing.
• Change the 'Door Phone Ring Timer', if required. This is the time for which the Door Phone will ring on the
extensions programmed as the landing destination. This Timer has a range of 001 to 255 seconds. By
default the Timer is set to 30 seconds.
• Repeat the same steps to program another door phone. If you have no other door phone to program, click
'Next' to navigate further.
• On this page, the Wizard displays the number of trunk ports you selected in the 'System Pre-Requisites'
page.
• Cost Factor is used to grade the cost of routing calls from that trunk, from 1 to 99; 1 signifies least cost and
99 signifies the highest cost. By default all trunks are assigned Cost Factor 01.
• Now configure the number of or part of the number and the preferred trunk to route that number. The
number can be a maximum of 16 digits.
• Select the cost factor trunk applicable for each number in the order of preference. Select the foremost
preferred Cost Factor trunk in Preference 1, the second most preferred Cost Factor trunk in Preference 2,
the third preferred Cost Factor trunk in Preference 3 and the least preferred Cost Factor trunk in
Preference 4.
ETERNITY supports four different types of LCR, but the Wizard supports only Number based LCR. To
know more, refer “Configuring LCR”.
Call Pick-Up allows extension users to answer (internal and trunk) calls ringing on other extensions from
their own extension; without physically going to the ringing extensions. For this extensions must be
assigned to CPU Groups.
• While you can create as many as 01 to 99 CPU Groups, and assign extensions to these groups, the
Wizard allows you to create and assign extensions to 01 to 16 CPU Groups. By default, all extensions are
assigned to CPU Group number 99.
• To create a CPU group, click the CPU group number, for example CPU 01, and select the extensions to be
assigned to this group from the box.
• If you have finished programming CPU groups, click 'Next' to navigate to the next page of the Wizard.
To know more about this feature, refer the topic “Call Pick Up”.
TWT Trunks
The Wizard makes configuration of the TWT Trunks easy with 'TWT Profiles'. Instead of configuring each
TWT Trunk individually, you can group together trunks that are to be assigned the same features - Calling
Line Identification, Trunk Landing Group - in a single TWT Profile.
You can program as many as 8 different TWT Profiles using the Wizard.
It is also possible to name each TWT Profile by the service provider. For example, you may rename TWT
Profile-1 created as 'BSNL', TWT Profile-2 as 'Reliance' and so on.
• To change the label of the TWT Profile, click the 'Pencil' icon.
• The name you entered will appear in place of 'TWT Profile' number.
• To assign the TWT Profile to the desired trunks, double click the field Select Trunks/Apply to Trunks.
You can select the trunks for the TWT Profile after configuring the other parameters of the TWT Profile.
• Define the Calling Line Identification Format for the TWT Profile by selecting the desired option in the box.
You are advised to consult the service provider in this regard.
• In the field 'Route Calls during Day to', select the landing destination for calls on trunks in the TWT Profile
during day time. You may select any option: Operator extension, other extensions, built-in Auto Attendant
or the Voice Mail Auto Attendant (if available) from the box.
• If you selected the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance
Messages) will be applied. Refer “Voice Message Applications” to know more.
• If you click the option 'Extensions' as the landing destination, select the extensions in the corresponding
field. Double click the field, the list box will open. Select the extensions from the left box.
A maximum of 32 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
• Click 'Submit' to save your settings in the TWT Profile. Repeat these steps to create another TWT Profile.
• If you have finished creating TWT profiles and assigning them to trunks, click 'Next' to navigate further.
• If there is any trunk you have not assigned a TWT Profile, the Wizard will pop-up an alert informing you
about the trunk you have not programmed and ask if you want the default TWT Profile number should be
assigned.
• If you wish to assign the default TWT Profile to such trunks, click 'OK'.
• Click 'Cancel' if you want to ignore the alert and not assign the profile to the TWT.
Do not make any modifications to them when configuring the system from the Full Programming Access
mode.
• You will reach this page only if your system has detected the presence of BRI Trunks or if you have
specified the number of BRI Trunks to be used in the 'System Pre-requisites' page.
• The ETERNITY supports a maximum of 32 BRI Trunks, depending on the system capacity of your
model.
• Using the Wizard, you can configure only the first four BRI Trunks in your system. To program the
remaining BRI Trunks, you must enter Full Programming Access.
The name will be displayed only if you have programmed names in the 'Naming of Trunks' page of the
Wizard. If you have not named the trunk on that page, you may change the label; right click the tab 'BRI
Trunk 1' and enter the desired name.
If required, you may change the Hardware slot and port number of the BRI trunk is displayed here.
• Select the mode for routing incoming calls. Incoming calls may be routed according to MSN Numbers, DDI
Numbers or Port-wise.
• Route Incoming Calls MSN Number wise: Select this option if you want to route incoming calls
according to MSN numbers66. You can program maximum 6 MSN numbers.
• Route Incoming Calls DDI Number wise: Select this option if you want to route incoming calls
according to DDI Numbers67. Select this option only if your extensions are arranged sequentially. If the
extensions are not arranged sequentially (for example, DDI numbers 2630551 to 2630559 are to be
routed to extensions 3001 to 3009) you are recommended to program this parameter from the Full
Programming Access mode. You can program maximum 12 DDI numbers.
• Route Incoming Calls Port wise: Select this option if you want to route all incoming calls on the BRI
trunk port to extensions, irrespective of dialed MSN/DDI number.
• MSN Number: Enter the MSN Number provided by your Service Provider. This number is used to
route the incoming calls and sent to the ISDN Exchange when making an outgoing call. You can enter
up to 6 MSN numbers using the Wizard.
• Route calls during day to: Select where you want to the route the calls during day time. You can route
the calls to the Operator or to a group of extensions or to Built-In Auto-Attendant or to the Voice Mail
Card's Auto-Attendant68.
If you click the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance
Messages) will be applied by the Wizard. Refer “Voice Message Applications” to know more.
66. MSN numbers is a set of numbers with no logical connection between the numbers themselves. For example, MSN numbers for a
BRI connection could be 2630555, 2634872, 2635098, etc. Up to 8 MSN numbers are provided on a single BRI connection.
67. DDI numbers are a set of numbers arranged sequentially, for example DDI numbers for a BRI connection could be 2630551 to
2630559.
68. The Built-In Attendant offers 5 simultaneous calls, whereas Voice Mail's Auto-Attendant offers 16 simultaneous calls.
A maximum of 32 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
• When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
• When Busy?: Select an option for what action the system should take when the landing destination is
busy. By default, it is set to 'Disconnect'.
• Route calls during night to: Select where you want to the route the calls during night time. If you
selected group of extensions as the destination, select the extensions from the list box, as described
above. This may be a different group of extensions than the one selected for calls during day time.
If you selected 'Extensions', you must now select the extensions by clicking the field provided for it. A
maximum of 32 extensions can be selected. Follow the same procedure described above for selection
of extensions for calls during day time.
• When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
• When Busy?: Select an option for what action the system should take when the landing destination is
busy. By default, it is set to 'Disconnect'.
• The Wizard allows you to select the desired landing extension (Operator/ extensions/ Built-In Auto
Attendant/Voice Mail's Auto Attendant) only for the first 3 MSN numbers. The remaining 3 MSN
numbers can be routed to a single extension only.
• So, if you want to route incoming calls to a particular extension during the day time and night time,
enter the MSN numbers in MSN#4 to MSN#6.
• If you selected DDI Number-wise routing, configure the following parameters for each DDI#:
• Root DDI #: Enter the Root DDI number. This number is used to send the DDI number to the ISDN
Exchange. This number is also sent to the Exchange when an outgoing call is made by an extension
which is not assigned DDI number. This is the same as the Pilot Number.
• Start DDI #: Enter the first DDI number provided by your Service Provider. More often than not, the
Start DDI# is the same as the Root DDI#.
• Total DDIs: Enter the total DDI numbers provided by your Service Provider.
For example, your Service Provider has given you DDI numbers 2630550 to 2630560. These are to be
assigned to extension numbers 2001 to 2010. In this case 2630550 will be the Root DDI# as well as
Start DDI#. As 10 DDI numbers are provided to you, the Total DDI# would be 10, and the Start
Extension# would be 2001.
• When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
• When Busy?: Select an option for what action the system should take when the landing destination is
busy, during DDI Ring Timer (default: 45 sec.). By default, it is set to 'Disconnect'.
• Route Calls during Day to: Select the landing destination for calls on BRI trunk during day time. You
may select any option: Operator extension, other extensions, built-in Auto Attendant or the Auto
Attendant of the Voice Mail System (if available).
If you click the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance
Messages) will be applied by the Wizard. Refer “Voice Message Applications” to know more.
If you selected 'Extensions' as landing destination, you must select the extensions. Double click the
field and select the extensions from the left box.
A maximum of 32 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
• Route Calls during Night to: Select the landing destination for calls on BRI trunk during night time.
Follow the same procedure for selecting extensions as described above.
• Click 'Submit' to save changes. Repeat the same steps to program the other BRI Trunks.
• If you have finished configuring the BRI Trunks, click 'Next' to navigate further.
• If there is a BRI trunk you have not configured, the Wizard will pop-up an alert informing you about the
trunk you have not programmed. It will ask you if you want the default BRI Profile number to be assigned to
the trunk.
If you wish to assign the default profile to such trunks, click 'OK'.
Click 'Cancel' if you want to ignore the alert and not assign the profile to the BRI trunk.
You may either configure this trunk by clicking 'Yes', or ignore this alert by clicking 'No', and navigate
further.
When programming the system from the Full Programming Access mode, do not modify the settings of
these resources. It will affect the settings made by the Wizard.
T1E1PRI Trunks
• The ETERNITY supports a maximum of 8 T1E1PRI Trunks, depending on the system capacity of your
model.
• However, using the Wizard you can configure only the first 2 PRI Trunks in the system. You may
configure the remaining Trunks, if applicable, from the Full Programming Access mode only.
• Change the label of the T1E1PRI Trunk, if required. Click the 'Pencil' icon. A prompt will pop-up. Enter the
desired name in the field of the prompt and click 'OK'. The name you programmed will appear.
You may enter the name of the Service Provider to make the identification of the Trunk easy.
• The Wizard will display the Hardware Slot-Port Number of the first T1E1 PRI Trunk the system has
detected in this field.
• Select the Signaling Type as applicable: from PRI, RBS, QSIG, and E&M. These are the signaling types
supported by the ETERNITY.
• Select the ISDN Switch variant. By default 'ETSI NET5' is selected as the variant.
• Select the mode for routing incoming calls. Incoming calls may be routed according to DDI Numbers or
Port-wise.
• Route Incoming Calls Port wise: Select this option if you want to route all incoming calls on the
T1E1PRI trunk port to groups of extensions, without identifying the DDI number.
• Route Incoming Calls DDI Number wise: Select this option if you want to route incoming calls
according to DDI numbers. Select this option only if your extension numbers are arranged sequentially.
If the extensions are not arranged sequentially you are recommended to program this parameter from
the Full Programming Access mode.
If your installation uses multiple DDI blocks on the same T1E1PRI connection, you can configure only 2
such DDI blocks using the Wizard. To configure more DDI blocks on a single T1E1PRI connection, you
must use the Full Programming Access mode.
• Root DDI #1: enter the Root DDI # provided by your Service Provider. The Root DDI # is the main
number assigned to the T1E1PRI trunk. It is also known as MSN Number, Pilot Number or Main
Number69.
If your exchange requires area code to be sent with the DDI number, program the root number with
area code. For example: Root DDI # is 2630555 and area code where the PBX is installed is 265, enter
2652630555 in this field.
• Route calls during day to: Select where you want to the route the calls during day time. You can route
the calls to the Operator or to a group of extensions or to Built-In Auto-Attendant or to the Voice Mail
Card's Auto-Attendant70.
If you click the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance
Messages) will be applied by the Wizard. Refer “Voice Message Applications” to know more.
If you selected 'Extensions', then select the extensions by clicking the empty field. A multiple selection
box will open. Select the extensions from the left box.
A maximum of 32 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
69. This number will be used to prepare the DDI number to the Exchange (Reverse DDI) when an Outgoing call is made from the
ETERNITY by the Extension assigned DDI Number. Also, this number will be sent to the Exchange without any modification when
an Outgoing call will be made by an extension which is not assigned DDI number.
70. The Built-In Attendant offers 5 simultaneous calls, whereas Voice Mail's Auto-Attendant offers 16 simultaneous calls.
• When Busy?: Select an option for what action the system should take when the landing destination
is busy (for the duration of the DDI Ring Timer; default: 45 seconds). By default, it is set to
'Disconnect'.
• Route calls during night to: Select where you want to the route the calls during night time from the
options. If you selected group of extensions as the destination, select the extensions from the box, as
described above. This may be a different group of extensions than the one selected for calls during day
time.
If you selected 'Extensions', you must now select the extensions by clicking the field provided for it. A
maximum of 32 extensions can be selected. Follow the same procedure described above for selection
of extensions for calls during day time.
• When NR?: same as described above for calls during day time.
• When Busy?: same as described above for calls during day time.
• Start DDI#: Enter the first DDI number given by the Service Provider. The Start DDI Number may be
the Root DDI number, as seen in most of the cases.
• Total DDIs: Enter the Total DDI numbers. Ask your Service Provider.
• Start Extension#: Enter the number of the first extension from which the DDI assignment is to be
done.
For example, your Service Provider has given you DDI numbers 2630551 to 2630560. These numbers
are to be routed to extensions 2001 to 2010. In this case, the Root DDI number as well as the Start
DDI# will be 2630550. As 10 DDI numbers are used, enter Total DDI numbers =10 and Start Extension
# = 2001.
• When NR?: same as described above for calls during day time.
• When Busy?: same as described above for calls during day time.
• Route Calls during Day to: Select the landing destination for calls on T1E1PRI trunks during day
time. You may select any option: Operator extension, other extensions, built-in Auto Attendant or the
Auto Attendant of the Voice Mail System (if available).
If you click the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance
Messages) will be applied by the Wizard. Refer “Voice Message Applications” to know more.
If you selected 'Extensions' as landing destination, you must select the extensions. Double click the
field and select the extensions from the left box.
The selected extension numbers will appear in the field, separated by comma.
• Route Calls during Night to: Select the landing destination for calls on T1E1PRI trunks during night
time. Follow the same procedure for selecting extensions as described above.
• Click 'Submit' to save changes. Repeat the same steps to program the second T1E1PRI Trunk.
• If you have finished configuring the T1E1 PRI Trunks, click 'Next' to navigate further.
Mobile Trunks
• You will reach this page only if your system has detected the presence of the Mobile Card in the system
(as the 'On-Site Configuration?' flag is enabled) or if you have specified the number of Mobile Ports
Used earlier in the 'System Pre-requisites page of the Wizard.
• ETERNITY supports up to 64 Mobile Ports, depending upon the system capacity of your model.
• The Wizard makes configuration of the Mobile Trunks easy with 'Mobile Profiles'. Instead of configuring
each Mobile Trunk individually, you can group together trunks that are to be assigned the same
features in a single Mobile Profile.
You can program as many as 4 different Mobile Profiles using the Wizard.
• Select the mobile trunks on which this Mobile Profile is to be applied. It is also possible to configure all the
other parameters and then select the mobile trunks.
To select the desired mobile trunks, double click the box. The Wizard will display the number of Mobile
Trunks you specified in the 'System Pre-requisites' page on the left side box. All the Mobile Trunks appear
in this box and are arranged sequentially in the increasing order of their software port number.
Place your cursor on the desired trunks and click 'Select' button. The selected Mobile Trunks will appear
on the right side box. It is also possible to select a range of trunks at a time pressing the 'SHIFT and 'Down'
arrow keys.
Recall that you have changed the SIM PIN of the SIM Card using a Handset before installing it in the
system to the default '1234'. Now, you may change it to a desired SIM PIN.
Since you are changing the SIM PIN for a Mobile Profile, this SIM PIN will be applied for all the Mobile
Trunks you selected for this Mobile Profile.
• Select the extensions to which the incoming calls on the Mobile Trunks should be routed.
If you select the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the default
Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance Messages) will
be applied by the Wizard. Refer “Voice Message Applications” to know more.
If you click the option 'Extensions' as the landing destination, select the extensions in the corresponding
field. Double click this field, the multiple selection box will open. Select the extensions from the left side
box.
A maximum of 32 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using drag and drop option.
The selected extension numbers will appear in the field, separated by a comma.
• Click 'Submit' to save your settings in the Mobile Profile. Repeat these steps to configure another Mobile
Profile.
• If you have finished configuring the Mobile Profiles, click 'Next' to navigate further.
• If there is any mobile trunk you have not assigned a Mobile Profile, the Wizard will pop-up an alert
informing you about the trunk you have not programmed. It will ask you whether the default Mobile Profile
number should be assigned to that trunk.
You may either assign a profile to this trunk by clicking 'OK', or ignore this alert by clicking 'Cancel, and
navigate further.
VoIP Network
• You will reach this page only if your system has detected the presence of the VoIP Card in the system
(as the 'On-Site Configuration?' flag is enabled).
• In case the On-site Configuration?' flag is disabled, you must specify 'Number of VoIP Ports Used' in
the 'System Pre-requisites page of the Wizard to be able to reach this page.
• Regardless of the number of VoIP Ports supported by your model of ETERNITY and the number of
VoIP ports you have defined in the 'System Pre-requisites' page, the Wizard allows you to configure
only the first four VoIP ports in your system.
• Configure the Network Parameters of each VoIP (WAN) Port. The system will automatically detect the
Hardware Slot and Port Offset and display the same for each VoIP Port.
• Change the label of the VoIP (WAN) Port, if required. This will make identification of the port easy.
• Click the 'Pencil' icon and enter the name of your choice in the blank field of the prompt that will pop up on
your screen. Click OK.
• In the Network Connection Type box, select the IP addressing scheme of the network to which the VoIP
(WAN) Port is connected: Static, DHCP, PPPoE. Ask your LAN administrator for this information. Select
the appropriate radio button.
• If your network connection is DHCP, skip the next two steps and select the DNS connection type.
The DHCP server on your network will automatically assign the IP Address and Subnet Mask, Gateway
Address and other parameters to the VoIP Port.
• If your network connection is Static, enter the following information from your LAN Administrator
• If your network connection is PPPoE, enter the following information from your Service Provider:
• PPPoE User Name
• PPPoE Password
• PPPoE Service Name (if applicable)
Select 'Static' to assign DNS IP Address and DNS Domain Name manually. If the Network Connection
Type is selected as Static, the DNS connection type can be selected as Static only.
Select 'Auto' to get DNS IP address and DNS Domain Name from PPPoE Server/DHCP Server. You can
select DNS Connection Type as Auto only if you select Network connection type as PPPoE or DHCP.
• Enter the Router's Public IP Address. This address will be used in SIP messages if the parameter 'Source
Port IP Address' in the page 'Configuring SIP trunks' is configured as Router's Public IP Address.
STUN server facilitates traversing through most NATs, except symmetric NATs. If your router has
symmetric NAT, do not program this field.
• Click to enable the check box 'Use SIP Port fetched using STUN?' if you want to use the port number
fetched using STUN in the SIP message.
This parameter should be disabled if you are using Port-Forwarding in the Router for SIP messages.
• Enter SIP UDP Port. This is port on which the ETERNITY (VoIP Port) listens for SIP messages. The VoIP
Port also uses this port to send SIP messages to the remote peer. The default value of SIP UDP Port is
5060.
• Enter RTP Listen Port. This port defines the port on which ETERNITY (VoIP Port) listens for RTP packets.
The system also uses the port as source port for sending RTP packets to the remote peer.
• Click 'Submit' to save the configuration of the VoIP (WAN) Port. Now, repeat the same steps to configure
another VoIP (WAN) Port.
• If you have finished configuring all VoIP ports, click 'Next' to navigate to further.
SIP Trunks
• You will reach this page only if you have specified 'Number of SIP Trunks Used' on the 'System Pre-
requisites' page.
• SIP Trunks are to be configured only if you are using Internet Telephony Service Providers for VoIP
calls.
• The number of SIP Trunks supported by ETERNITY varies according to its model. ETERNITY ME
supports 32 SIP Trunks. ETERNITY GE supports 16 SIP Trunks and ETERNITY PE supports 4 SIP
Trunks.
• The Wizard however, allows you to configure only 4 SIP Trunks (even if you have specified more than 4
SIP Trunks being used on the 'System Pre-Requisites' page). To program the remaining IP Trunks, you
must use the Full Programming Access mode.
• ETERNITY supports Fax over IP (FoIP). The Wizard allows you to configure the Fax Type only. To
configure Fax Homing, you must enter the Full Programming Access mode.
• Assign a VoIP Software Port to this SIP trunk in the field 'VoIP S/w Port'.
• The Wizard will display the Hardware Slot and the Port number of the first VoIP Card detected in the field
'H/w Slot - Port Number'.
If multiple VoIP cards are installed in the system, you may change the Hardware Slot and Port number as
desired.
• You may change the label of the SIP Trunk (rename 'SIP Trunk 1'), if required. You may use the name of
the ITSP to make identification of the SIP Trunk easy.
Click the 'Pencil' icon. A prompt will pop up, asking you if you want to change the name of this SIP Trunk.
Enter the desired name in the blank field. Click 'OK'. The new name will appear instead of the SIP Trunk
number.
• Enter the SIP ID provided by the ITSP. This is the ID which callers will use to call this SIP Trunk. The SIP
ID may be a number or text.
• Enter the Proxy/Registrar Server Address and the Registrar Server Port provided by the ITSP. The
Registrar Server Address can be an IP Address or domain. The Registrar Server Listening Port ranges
from 1024 to 65535. The default Registrar Server Port is 5060.
• Enter the Authentication ID (User ID) and Password provided by the ITSP.
• Enable 'Outbound Proxy' by selecting the checkbox, if the ITSP who provided this SIP Trunk has a SIP
outbound server to handle voice calls. By default 'Outbound Proxy' is disabled.
• If you have enabled 'Outbound Proxy', enter the Outbound Proxy Server Address and the Server Port
provided by the ITSP. This can be an IP Address or Domain name.
• Set the desired Vocoder Preference for this SIP Trunk. Vocoders are the various Voice Codecs used to
compress the data in RTP packets for optimum use of bandwidth and for ensuring voice quality. You can
set 7 Vocoders options in the order of preference for this SIP account.
• Select the DTMF Option. The DTMF option you select will determine how the DTMF digits will be sent over
the IP Network, when a DTMF digit is pressed. ETERNITY VoIP Card supports three DTMF options: RTP
(RFC 2833), SIP Info, and InBand. Select the appropriate option. By default RTP (RFC 2833) is selected.
• Select the appropriate Fax Type. The ETERNITY VoIP card supports T.38 (UDPTL), T.39 (RTP) and Pass-
Through.
• Select 'Use Ethernet Port IP Address' if the VoIP port (of the ETERNITY VoIP Card) is connected directly
to the public internet.
• Select 'Use IP Address Fetched Using STUN' if the VoIP port (of the ETERNITY VoIP Card) is located
behind a NAT router other than Symmetric.
• Select 'Use Router's Public IP Address', if the VoIP port (of the ETERNITY VoIP Card) is located behind a
NAT Router (any type).
• In 'No. of Simultaneous Calls', select the number of simultaneous calls you want to allow on this SIP Trunk.
The number of simultaneous SIP calls depends on the number of simultaneous calls allowed by the ITSP
with whom you have subscribed this SIP Trunk and the number of simultaneous calls supported by your
model of ETERNITY.
• In 'Route Calls during Day to', select the landing destination for calls on SIP trunks during day time. You
may select any option: Operator extension, other extensions, built-in Auto Attendant or the Auto Attendant
of the Voice Mail System (if available).
• If you click the option ‘Built-In Auto Attendant’, the default Voice Modules 02 to 13 containing the default
Voice Messages (Morning, Afternoon and Evening Greetings, DID Greeting and Guidance Messages) will
be applied by the Wizard. Refer “Voice Message Applications” to know more.
• If you click the option 'Extensions' as the landing destination, select the extensions in the corresponding
field. Double click the field, the list box will open. Select the extensions from the left box.
A maximum of 32 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
• In the field 'Route Calls during Night to', select the landing destination for calls on SIP trunks during night
time. Follow the same instruction as in the previous step to select extensions as the landing destination.
• Click 'Submit' to save your settings in the SIP Trunk. Repeat these steps to configure another SIP Trunk.
• If you have finished configuring the SIP Trunks, click 'Next' to navigate further.
• If there is any SIP Trunk you have not configured, the Wizard will pop-up an alert informing you about the
trunk you have not programmed.
You may either program this trunk by clicking 'Yes', or ignore this alert by clicking 'No', and navigate
further.
For the SIP Trunks, the Wizard uses the following resources:
• Routing Groups 67-74.
• Trunk Feature Template 46-49.
• SIP Hardware Template 2-5.
Do not change the these resources when using Full Programming Access mode.
Emergency Numbers
• This page displays the default Emergency Numbers according to the 'Region' selected for the system.
In other words, it displays the Emergency Numbers specific to your country where the ETERNITY is
installed.
• The Emergency Numbers on this page are non-editable. All you need to do is to select the Outgoing
Trunk Bundle Group (OGTB) for each Emergency Number. For example, '112' is the default
Emergency Number for the mobile network. So, you may select the Mobile Trunk for dialing this
number.
• For each default Emergency number, select the trunks to be used to route this call. Double click the
'Through' field. The box will open, displaying the trunk types present in the system in the left box.
All the Trunk types present in the system are arranged by their type, in the sequence of their software port,
irrespective of their hardware location.
• Double click the desired trunk type and click 'Select'. The selected trunk type will appear on the box on the
right.
If you select trunks of the same type in sequential order, like: TWT001, TWT002, TWT003, BRI01, BRI02
and MOB02, the same trunk type will be grouped in one OG Trunk Bundle; TWT001, TWT002, TWT003
will be OGTB #1, BRI01 and BRI02 will be grouped as OGTB#2, and MOB02 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, for example: TWT001, MOB01, BRI01
and TWT002, four OGTBs will be formed with TWT001 as member of OGTB#1, MOB01 as member of
OGTB#2, BRI01 in OGTB#3 and TWT002 in OGTB#4. So, despite two same trunk types being selected
(TWT001 and TWT002) they are grouped in separate OGTBs.
It is possible to change the sequence of trunks on the right side box using the drag and drop action. You
can also delete a particular trunk from the right box.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
• If you selected a Region other than the United States or Australia, you may get the message “No
Emergency numbers programmed”.
• You may program the Emergency Numbers of your country using “Full Programming Access” mode.
Disclaimer:
• Matrix Comsec will not be responsible for incorrect programming of Emergency Numbers.
Done
You have now reached the last screen of the Wizard. Click 'Done' to submit the configuration you have done using
the Wizard.
When you click 'Done', the Wizard will throw up alerts for incomplete or missing configuration information, for
example, if you have not assigned a User Profile to an extension, or if you have not assigned a TWT Trunk Profile
to a TWT Trunk, if you have configured a BRI or T1E1PRI Trunk.
After completing the validation of the configuration, the Wizard will inform you about successful configuration of the
system with the message: “Congratulations! System is configured as per your wish".
Each page of the Wizard also has an 'Exit' button, which you may click to exit the Wizard.
However, the changes you made till the page you exited from will not be applied to the system configuration files.
• The Standard PBX Wizard has been designed keeping a very broad user base in mind and is limited to
the configuration of the parameters most commonly required by a broad user group. So, it does not
cover the configuration of all the features and facilities of the system.
• The Wizard does not cover the configuration of the following features and facilities. These are to be
configured from the Full Programming Access mode only.
• Global Directory
• Personal Directory
• Account Name List
• Automatic Number Translation
• Automated Control Applications
• Call Cost Calculation
• Call Duration Control
• CLI-based Routing
• Closed User Groups
• Communication Port
• Department Groups
• Digital Input Port
• Digital Output Port
• DISA CLI Authentication
• Digital Key Phone Key Template
• DND Text messages
• DST Parameters
• E&M Parameters
• E&M Feature Templates
• ISDN Terminal Parameters
• Keyboard Macros
• Logical Partition
• Master Ethernet Parameters
• Page Zones
• Peer-to-Peer table
• Ring parameters
• RTC parameters
• Security Alarm Parameters
• Station Message Detail Records
• System Activity Log
• System Fault Log
• System General Parameters
• System Timers and Counts
Full Programming Access allows configuration or all programmable parameters of the system.
You can enter Full Programming Access using Jeeves or an extension Telephone (DKP or SLT) of the ETERNITY.
You are recommended to use Jeeves. If you choose to use a telephone, you are recommended to use a DKP for
ease of operation.
To enter Full Programming Access mode using Jeeves by logging in as “System Engineer Login”, and clicking the
'Use Full Programming Access' link.
To enter Full Programming Access using a Telephone, enter into SE mode from a DKP/SLT by dialing 1#91
followed by the SE password (default: 1234).
For detailed instructions, refer the topic “Entering the SE mode using a Telephone”.
Avoid modifications or use of the following system resources, when configuring the system from the Full
Programming Access mode.
• Station Basic Feature Template number 2 to 10.
• Class of Service Groups 2 to 19.
• Outgoing Trunk Bundle Groups 2 to 27
• Outgoing Trunk Bundles 5 to 108.
The system uses these resources for the Quick Installation Wizard-Standard PBX. Changes made in them will not
be updated in the Wizard.
System resources (number of trunk and extension ports supported) vary by models and variants of ETERNITY. It is
quite common for users to utilize the system resources below its capacity, especially when they begin using a new
system.
To make the task of configuring for such users easier, ETERNITY allows you select the 'Model Type' you are using,
and specify the number of trunk and extension ports you want to configure. Accordingly all the relevant pages of
Jeeves will show only as many trunk and extension ports that you have specified, instead of showing all the ports
supported by ETERNITY.
To be able to do this, you must configure the 'System Pre-requisites' using Jeeves or a Telephone.
On-site Configuration
ETERNITY makes configuration even more focused by making it possible to configure only those trunk and
extension ports which are actually present in the system.
When 'On Site Configuration' flag is enabled, ETERNITY will detect all the different trunk and extension port types
present in the system (at Power-ON). Accordingly, all the relevant pages of Jeeves will show only those ports
detected by the system for configuration.
The system will detect the presence of ports at each Power ON/Reset. Whenever a new card is found, the range of
ports is updated and displayed on Jeeves. You can then define the number of ports to be used.
When the On Site Configuration' flag is enabled, the Quick Installation Wizard - Standard PBX will also
display only the ports and port types that are on-board. Enable this flag if you want the Wizard to display
only the port types present in the system.
It is recommended that you enable the 'On-Site Configuration?' flag when you are configuring the system
at the installation site.
• Model Type: Select the model of ETERNITY you are configuring from this list.
When you select the 'Model Type', Jeeves will display the different port types according to the system
capacity (the maximum number of ports) supported by the model and variant of ETERNITY you have
selected as the Model Type.
For example, if you selected ETERNITY GE6S as Model Type, the maximum number of TWT Ports will
be 96 and the maximum number of DKP extensions would be 96 ports. If you selected ETERNITY
PE6SP, the maximum number of TWT Ports will be 16, and the maximum number of DKP ports will be
48.
• Number of Ports Used: Define the number of ports to be used for each Port Type (TWT, DKP, SLT,
Mobile, T1E1 PRI, BRI, VoIP, SIP) in the respective boxes.
For example, if you want 8 TWT Trunks, 24 DKP extensions, and 128 SLT extensions to be used,
select the same numbers in the respective boxes.
If you want to use voice mail, or connect any device to the Digital Input Port or the Digital Output port,
select 'Yes'.
• On Site Configuration: If you want to enable 'On-site Configuration' flag, click the check box.
Click 'Submit' at the bottom of the page. OR click away on any other part of your screen.
When you enable this flag, the fields for 'Number of Ports Used', will be populated with exactly the
number the system has detected.
For example, the system has detected ETERNITY ME SLT32 card at power-on, so the maximum
number of SLT ports in the will be 32. If you want only 20 SLTs to be used, select 20.
To cite another example, if your system is ETERNITY PE3SS, which does not support ISDN BRI or
PRI, the fields of these port types and ISDN Terminals on this page are displayed as non-editable.
Similarly, the field of the Door Phone ports will be editable only if the system has detected a Door
Phone card.
• If the system has detected BRI ports, the fields 'BRI Trunks used' and 'BRI ISDN Terminals used' will
be editable.
• By default, the number of BRI Trunks Used will be equal to the Number of BRI ports used.
• The number of ISDN terminals used will be zero. Only when the System Engineer changes the value of
the Number of BRI Trunks used will the number of ISDN Terminal get changed to the number of
available BRI ports x 8. In other words, the number of ISDN terminals will be: Number of BRI Ports
used minus the Number of BRI Trunks used multiplied by 8.
The ETERNITY is a versatile system that can operate anywhere in the world, meeting the diverse customer
requirements worldwide.
To speed up the process of system configuration, ETERNITY is supplied with factory-set values for system and
feature settings, referred to as “Default Settings”. These factory-set values are loaded when the system is installed
and are sufficient for getting the system into operation. However, users may alter or customize the Default Settings
to match their exact requirement.
ETERNITY provides Default Settings to match country/region-specific requirements of users around the world. The
system is designed to work efficiently in any country with these default settings.
To load the country-specific Default Settings, users must select the Region for the country in which the system is
installed.
India is selected as the default Region. So, if you are installing ETERNITY in a country other than India, change the
Region.
• In the Region list, select the country where the system is installed.
• To load the country-specific default settings, click the link 'Default the System'.
• You will get an alert that default values will be assigned and ask you if you want to continue.
• Click 'OK'.
• Exit SE mode.
The Master Ethernet Port of ETERNITY allows you to connect ETERNITY to a standalone PC or a LAN PC to
operate Jeeves, capturing Station Message Detail Records, generating System Activity Logs and Reports, and
running the Property Management Software (PMS) in hotels, and Call Accounting Software.
When you connect ETERNITY to a LAN switch, the IP Address of the Master Ethernet Port of ETERNITY must not
conflict with the IP Address assigned to any device on the LAN, and the Master Ethernet Port of ETERNITY must
be in the same Subnet as the LAN PC from which it is to be accessed.
The Master Ethernet Port can be connected to a LAN Switch/Hub/Broadband Router/Modem. Since Network
scenario may vary from site to site of the installation, you must accordingly configure the Network parameters of the
Master Ethernet Port. This can be done using Jeeves and by dialing SE commands from a telephone.
• Enter the IP address of Master Ethernet Port of ETERNITY on the address bar or the browser.
You may enter the default IP Address of the Master Ethernet Port: 192.168.1.101
OR
If the IP has been changed, enter the current IP Address of the Master Ethernet Port.
• The 'Welcome' page will open, with links to the different Login modes.
• Click the link 'Login as System Engineer (SE)', enter the System Engineer password.
The default password is 1234. If the password has been changed, use the current password.
• Click the link ‘Use Full Programming Access’ to open the page.
Select the IP Addressing scheme of your network: Static, DHCP, PPPoE. By default, Static is selected as
Connection Type.
• IP Address: Enter the IP Address you obtained from your network Administrator for the Master
Ethernet Port of ETERNITY in this field. Make sure that the IP Address does not conflict with that of
any other device on the LAN.
• Subnet Mask: Enter the Subnet Mask you obtained from your network Administrator for the Master
Ethernet Port in this field. When connected on a LAN, ETERNITY should be in the same Subnet as
the LAN PC from which it is to be accessed.
• Default Gateway: Enter the Router’s LAN Interface IP Address as the Default Gateway IP
Address.Now assign these to the Master Ethernet Port.
• PPPoE User ID: Enter the User Name provided by the Internet Service Provider. The User ID may be
a maximum of 16 characters.
• PPPoE Password: Enter the User Password provided by the Internet Service Provider. The password
may be a maximum of 16 characters.
• PPPoE Service Name: Enter the Service Name, if provided by your Internet Service Provider. The
Service Name may consist of a maximum of 16 characters. If Service Name is not provided, leave this
field blank.
Configure the DNS related parameters as provided by your Internet Service Provider. You may consult your LAN
administrator in this regard.
• DNS Address Assignment: If you selected DHCP or PPPoE as Connection Type and the DHCP/
PPPoE Server provides DNS Address, select ‘Auto’ as the DNS Address Assignment.
If you selected DHCP or PPPoE as Connection Type, but the DHCP/PPPoE server does not assign
DNS Address, then you must select ‘Static’ as DNS Address Assignment, and manually program the
Primary and Secondary DNS Server Address, as applicable.
If you selected Static as Connection Type, you can select only ‘Static’ as the DNS Address Assignment
and program the Primary and Secondary DNS Server Address, as applicable.
• Primary DNS Server: If you selected ‘Static’ as the DNS Address Assignment, enter the DNS Server
Address in this field. If your Internet Service Provider has provided you with a Primary and a Secondary
DNS Server Address, enter the Primary DNS Server in this field.
• Secondary DNS Server: If you selected ‘Static’ as the DNS Address Assignment, and your Internet
Service Provider has given you a Primary and a Secondary DNS Server, enter the Secondary DNS
Server address in this field. If your Service Provider has not given you any secondary DNS Server,
keep this field blank.
• If you have configured the above basic Ethernet Port parameters, click ‘Submit’ to save your settings.
Advanced Configuration
• To configure other advanced configuration parameters, click the ‘Advance’ button, and scroll with the
vertical bar to reach the desired parameter.
• Define the Listening Port for the Web Server of Jeeves in this field. The valid port range is 80, 1024 to
65535. By default, 80 is assigned as Web Server Port for Jeeves.
• Dynamic DNS makes it easy for users to access and use the GUI Jeeves from the public IP network by
using a Host Name instead of the IP Address of the Master Ethernet Port.
ETERNITY supports Dynamic DNS client of the Service Provider DynDNS.org. If you wish to use
DynDNS.org, program the following parameters:
• Enable Dynamic DNS: Enable this flag if you are using the services of DynDNS.org. By default, this
flag is disabled.
• User ID: Enter the User ID you created on DynDNS.org here. The User ID may be 24-characters long.
• Password: Enter the password you created on the DynDNS.org here. A maximum of 40 characters,
including all ASCII characters are allowed.
• Host Name: Enter the Host Name you created on DynDNS.org here. The Host Name may consist of a
maximum of 40 characters, including all ASCII characters.
• Update IP Address now?: When the Master Ethernet Port is registered with the Dynamic DNS server,
the server stores the mapping between the host name and the public IP address and updates this
whenever the Public IP address changes. However, it is possible to update the IP address at will, by
clicking this button.
• Router’s Public IP Address: This parameter is to be configured when Dynamic DNS is to be used in
the following scenario:
• the Master Ethernet Port of ETERNITY is connected behind a NAT Router.
• Static IP Addressing is used by the Internet Service Provider to assign Public IP to the Router.
If this is your installation scenario, enter the Router’s IP Address in this field. By default this field is
blank.
• Simple Traversal of UDP through NAT (STUN): This parameter is to be configured when Dynamic
DNS is to be used in the following scenario:
• the Master Ethernet Port of ETERNITY is connected behind a NAT Router.
• Dynamic IP Addressing (DHCP/PPPoE server) is used by the Service Provider to assign Public IP
Address to the Router.
The Master Ethernet Port will use STUN Query to know the current Public IP Address of the NAT
Router.
• STUN Server Address: Enter the STUN Server Address; a maximum of 40 characters are allowed.
• STUN Server Port: Enter the Listening Port of the STUN Server. The valid range for this field is
from 1024-65535. The default STUN Port is 03478.
• STUN Query Interval (min): This parameter defines the interval between each STUN query for the
Public IP Address of the NAT Router. The range of this interval is from 0001 to 9999 minutes. By
default, it is set to 120 minutes.
• If you have finished programming the Ethernet Port Parameters, log out of Jeeves or continue
programming.
To Update Dynamic DNS IP Address binding (‘Update IP Address Now?’ flag), dial
• 2130
• Exit SE mode.
• Exit SE mode.
• SLT Extension Ports: Single Line Telephones (SLT) is connected to these ports. ETERNITY supports up
to 51271 SLT extensions. The number of SLT ports available to you depends on the model and variant of
ETERNITY, and number and configuration of the SLT Cards installed in the system.
• DKP Extension Ports: The proprietary Digital Key Phone of Matrix is connected to these ports.
ETERNITY supports up to 12872 DKP extensions. The number of DKP ports available to you depends on
the model and variant of ETERNITY, and number and configuration of the DKP Cards installed in the
system.
• ISDN Terminals: These are ISDN phones connected to the BRI Ports of the ETERNITY. ISDN Terminals
can be connected only in a Point-to-Multipoint BRI configuration (Short or Extended Passive Bus
Configuration). Refer the topic Installing BRI Card under Installation instructions for your model of
ETERNITY to know more.
A maximum of 8 ISDN Terminals (phones) can be connected on a single BRI Bus line in a Point-to-
Multipoint configuration. In a Short Passive Bus Configuration, you can connect up to 8 ISDN Terminals
while in the Extended Passive Bus Configuration you can connect up to 3 ISDN Terminals.
Depending on the number of BRI ports available to you and the type of Point-to-Multipoint Configuration
(Short or Extended Passive Bus), a maximum of 64 ISDN Terminals can be connected to the ETERNITY.
• SIP Extensions: Any SIP-enabled device like an IP-phone, a Softphone, a Wi-Fi mobile handset, or a
PDA can be registered with the VoIP Card of ETERNITY and function as the 'SIP Extension' of the
ETERNITY.
SIP Extensions function like any normal DKP/SLT extension of the ETERNITY, allowing you to make and
receive calls to any extension user of the ETERNITY as well as any external numbers over PSTN, GSM,
VoIP and E&M lines, depending on the “Logical Partition” configured in the System.
ETERNITY ME supports 999 SIP Extensions. ETERNITY GE supports 500 SIP extensions and
ETERNITY PE supports 50 SIP Extensions. SIP Extensions are a licensed feature. To know more, refer
the topic “License Management”.
• Magneto Ports: Magneto telephones are connected to these ports and function as extensions of the
ETERNITY ME. A maximum of 128 Magneto Ports are supported.
• E&M Ports functioning as Stations: An E&M port of ETERNITY can be programmed to take on the
function of a Subscriber (Station), to work like a station interface, receiving incoming calls.
Presuming that you have connected the extensions successfully, you may now configure the extensions
using Jeeves or a Telephone.
71. The maximum number of SLT ports supported in ETERNITY ME may vary according to the type of Power Supply (whether PSUNI
or DC) being used. See Technical Specifications provided in the Appendix.
72. The maximum number of DKP ports supported in ETERNITY ME may vary according to the type of Power Supply (whether PSUNI
or DC) being used. See Technical Specifications provided in the Appendix.
• Station Basic Feature Template - for DKP, SLT, and SIP Extensions, ISDN Terminals, and E&M Ports
functioning as stations, and Magneto ports.
• Station Advanced Feature Template - for DKP, SLT, and SIP Extensions, ISDN Terminals, and E&M Ports
functioning as stations, and Magneto ports.
• SIP Hardware Template - for SIP Extensions (and SIP Trunks) only.
You can use these templates to program extensions which are to be assigned the same set of features at one go,
saving you the effort for painstaking configuration of each extension.
The features in these templates are loaded with default values that fulfill the requirements of a very broad user
base. The Templates may be customized as per user requirements and applied to the extensions.
Before you start the configuration of the extensions, please read the description of the templates and how to
customize the templates according to user requirements.
The SLT Hardware Template allows you to configure according to user requirements, a common set parameters
(features) like Caller ID Presentation (DTMF, FSK), Digit Pad Count, Ring Type, AC Impedance, Answer Signaling
type, etc. to be assigned to all SLT Hardware Ports.
Each of these hardware parameters, along with its default value, is briefly described below.
The Eternity supports 3 signaling protocols for CLI on the SLT port: DTMF, FSK-V.23, and FSK-BellCore.
Select the appropriate signaling protocol.
2. Digit Pad Count: Certain SLT instruments that support CLI require a minimum number of digits in the
calling party's number to be able identify and display it. The Digit Pad Count signifies the number of zeroes
to be added with the Calling party's number before displaying it on the called party's instrument. This count
entirely depends on the instrument connected to it.
3. Ring Type: The SLIC used with SLT port allows you to change the Ring type: Sinusoidal, Trapezoidal, Low
Sinusoidal, Low Trapezoidal. This is helpful in cases when telephone instruments, which expect sinusoidal
4. Tx Gain: The SLIC used with each SLT port provides a facility to adjust the Transmit (Tx) Gain. This
enables the SLT user to increase the volume of the outgoing speech on the SLT, if required.
5. Rx Gain: The SLIC used with each SLT port provides a facility to adjust the Receive (Rx) Gain. This
enables the SLT user to increase the volume of the incoming speech on the SLT, if required.
6. AC Impedance: The SLIC used with each SLT port provides a facility to adjust the AC impedance of the
SLT port with the communication equipment connected to it.
Generally, most telephone instruments that are connected have nominal characteristics with AC
impedance of 600. However, the ETERNITY allows you to connect instruments with AC impedance other
than 600.
7. Flash Timer (msec): In Pulse Dialing, codes are dialed in pulses. A Flash key is generally used to dial this
code. Flash is breaking the loop current for 83ms to 900ms. Flash Timer defines the time period which
should be considered as Flash, if the loop current breaks.
The range of the Flash Timer is from 83 to 999 seconds. By default, the Flash Timer is set to 600 msec.
Program the Flash Timer as per user requirement.
8. Answer Signaling: An 'Answer Signal' is a signal generated on the SLT port to indicate that the called
party (remote party) has answered the call and the call is now mature.
Answer Signaling on the SLT port is particularly useful when there is a PCO machine or any Billing
equipment connected to the SLT port. With Answer Signaling enabled on an SLT port, during an outgoing
call is made from that SLT port to any other port - TWT/Mobile/SIP/T1E1/BRI - when the called party
(remote party) answers, the Public Network provides an Answer Signal to the trunk port to indicate call
maturity.
This information can be passed on to the PCO machine billing equipment in the form of Answer Signaling.
On detecting Answer Signaling the PCO machine billing equipment can start billing.
Answer Signaling is generated in the form of Polarity Reversal or Battery Reversal, whereby the Battery
polarity of the SLT port gets reversed. For example, if the battery polarity of the SLT port is +ve for TIP and
-ve for RING in speech condition, then on call maturity, TIP becomes -ve and Ring becomes +ve.
To generate Answer Signaling on the SLT Port, select 'Polarity Reversal'. Select 'None' if Answer Signaling
is not be generated on the SLT port.
9. Disconnect Signaling: A 'Disconnect Signal' is the signal generated on the SLT port to indicate that the
called party (remote party has disconnected the call.
Disconnect Signaling on the SLT port is useful when there is a PCO machine or any Billing equipment
connected to the SLT port. With Disconnect Signaling enabled on an SLT port, during an outgoing call is
made from that SLT port to any other port - TWT/Mobile/SIP/T1E1/BRI - when the called party (remote
party) disconnects (goes ON Hook), the Public Network provides a Disconnect Signal to trunk port indicate
call disconnection. This signal can be generated on the on the SLT port to indicate to the PCO machine/
• Polarity Reversal: Call Disconnection is signaled in the form of Polarity Reversal. The Battery polarity
of the SLT port will be reversed. For example, if the battery polarity of the SLT port is '+ve' for TIP and '-
ve' for RING in speech condition then on disconnection on other port, TIP will become '-ve' and Ring
'+ve'. When call is disconnected, user will get Error tone.
• Open Loop: Call Disconnection is signaled in the form of Open Loop Disconnect Pulse, whereby the
Battery voltage on the SLT port is removed for the duration of the Open Loop Disconnect Timer
programmed for that SLT port and will be restored on the expiry of this Timer. However, the Polarity of
Battery Voltage on the SLT port is not changed. When call is disconnected, the SLT extension user gets
an Error tone.
To generate Disconnect Signaling on the SLT Port, select 'Polarity Reversal' or 'Open Loop' as
appropriate. Select 'None' if Disconnect Signaling is not be generated on the SLT port.
10. Open Loop Disconnect Timer (msec): This parameter is applicable only if the option Open Loop
Disconnect is selected as Disconnect Signaling type on the SLT port.
Open Loop Disconnect Timer is the time period for which the system will remove Battery Voltage on the
SLT port and restore Battery Voltage on the expiry of the Timer to signal Call Disconnection.
The range of this timer is from 001 to 999 milliseconds. By default, the Timer is set to 500 msec.
11. Loop Current (mA): The SLT Port provides Loop Current to the telephone instrument connected to the
SLT port to drive the telephone instrument.
The Loop Current is to be increased/decreased according to the length of the telephony wiring cable
between the wall jack (into which the SLT telephone instrument is plugged) and the MDF (into which the
cables from the SLT port are terminated).
The longer the Loop Length of the SLT port, the greater the likelihood of current dissipation, affecting
speech quality of the telephone instrument connected to the SLT port.
The system supports Loop Current of 25, 30, 35 and 40 mA. By default the Loop Current for SLT ports is
set to 25mA and the SLT port, which is sufficient to support Loop Length of 1 kilometer.
You may change the Loop Current according to the Loop Length of the SLT.
12. Minimum Current for OFF-Hook Detection: ETERNITY detects OFF-Hook state of an SLT instrument
and gives dial tone on the basis of the current drawn by it from the SLT port. However, all types and brands
of SLT instruments may not uniformly draw the same minimum current; some may draw lesser and some
may draw more, making OFF-Hook detection difficult for ETERNITY. To resolve this, ETERNITY provides
for programmable values for threshold current for OFF-Hook detection: 10mA, 12mA, 14mA, 16mA and
18mA.
By default, the value of the Minimum Current for OFF-Hook detection is set to 12 mA. Change this value
according to the current drawn by your SLT instrument.
13. ON-Hook Detection Current (mA): ETERNITY detects ON-Hook state of an SLT instrument to route calls
on the basis of the current drawn by it from the SLT port. However, as all types and brands of SLT
instruments may not uniformly draw the same current, ON-Hook detection becomes difficult for the system.
To resolve this, ETERNITY provides for programmable values for threshold current for ON-Hook
Detection: 10mA, 12mA, 14mA, 16mA and 18mA.
SLT instruments also vary in the level of current drawn during the normal 'idle' state and when Flash is
dialed73 (the simulated idle state). So, when the Flash key of an SLT instrument is pressed, and if the
instrument draws a higher current than the threshold defined for the 'idle' state, the system will not be able
to detect Flash (ON-Hook state).
Consider this when changing the value of ON-Hook Detection Current. Define the value considering the
current drawn by your SLT instrument in idle state, as well as when Flash key is pressed.
14. Rx Gain at SIP Trunk/SIP Extensions (dB): This parameter allows you to increase the incoming speech
volume level of calls from SIP trunks/SIP Extensions to SLT stations. By default, Rx Gain is set to 0dB.
15. Tx Gain at SIP Trunk/SIP Extensions (dB): This parameter allows you to increase the outgoing speech
volume level of calls from SIP trunks/SIP Extensions to SLT stations. By default, Tx Gain is set to 6dB.
• You can also set the Rx and Tx Gains for SIP to Digital Trunks and Stations and SIP to TWT Trunks.
• To increase Rx and Tx Gain for SIP to TWT trunks, go to 'TWT Hardware Template’.
• To increase Rx and Tx Gain for Digital Trunks/Stations, go to 'SIP Trunk Parameters’.
The default parameter values of the SLT Hardware Templates are country specific and are loaded in each template
according to the Country selected as the “Region”.
For example, when India is selected as the Region, the default value of the CLIP Type in the SLT Hardware
Templates is DTMF, whereas it is FSK-Bellcore when the Region is selected as US or Canada and FSK-V.23 when
UK is selected as Region. Similarly, the default values of AC Impedance on the SLT Hardware Templates will vary
according to the Region selected; 600 ohms for Region India, 900 for Region the Philippines, and 350 +(1000
|| 0.21µF) for Region UK.
By default SLT Hardware Template Number 01 is assigned to all the SLT ports. This template has default values
fulfilling the common requirements of a very broad user base.
73. Dialing 'Flash' either with the 'Flash Key' or by pressing the Hook-switch causes the phone to go in ON-Hook state briefly for 600-
800 milliseconds. Thus ON-Hook state is simulated briefly. The SLT may draw a higher current when 'Flash' is dialed.
If you want to change the values of certain SLT Hardware Parameters, but apply the same parameter values to all
SLT ports, simply customize the desired parameters in Template 01.
However, if different hardware parameters are to be applied to different SLTs, then you can customize different the
SLT Hardware Templates using Jeeves or a Telephone.
• Click the 'SLT Hardware Template' under SLT Configuration to open the page.
• Select a Template number you wish to customize, for example Template 02.
• Now, apply this SLT Hardware Template 02 on the SLT ports. To do this,
• Go to the SLT software ports to which this Template is to be assigned, for example SLT-001, 002, and 003.
For example, to change the CLIP Type in Template 02 from default the DTMF to FSK-Bell, dial 5702-1-
02-01-3
Where,
02 is the template number
01 is the parameter number for CLIP Type
3 is the code for FSK-Bell
• 'Cmplx' is 'Complex Impedance' which consists of resistive and reactive impedance for the 'AC
Impedance' parameter.
• The default values of the SLT Hardware Templates are for the default Region India. The default values
will differ according to the Region you have selected for the system.
When the SE command to set the AC impedance for the SLT is issued, the settings will be effective when
the SLT extension (for which AC Impedance is programmed) goes ON-Hook. ETERNITY will restart, if the
SLT is OFF-Hook.
• Exit SE mode.
The ETERNITY offers 50 such Station Basic Feature Templates. A Station Basic Feature Template is assigned to
all the types of stations, namely SLT, DKP, ISDN Terminals, E&M (Orientation = Station), T1E1PRI (NT mode) and
BRI (NT mode).
These templates have commonly used values, but can be customized per the requirement and applied on the
extensions.
• Time Table: A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours,
Non-Working hours for a week.
Certain features of the ETERNITY like Operator, Class of Service, Toll Control, Outgoing Trunk Access,
among others, require the station to behave differently in each Time Zone74.
So, a Time Table is assigned to stations defining the Time Zones for the entire week, so that the system
can execute the Time Zone-dependent features and facilities according to the Time Table.
There are 8 different Time Table templates to select from. By default, the Time Table 1 is assigned to all
Station Basic Feature Templates. All seven days of the week are 'working hours 9:00 to 18:00' with break
hours from '13:00 to14:00'.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the
Station Basic Feature Template. Please refer the topic “Time Tables” for more details.
• Operator: Define the Operator for the stations on which the template is applied.
The system supports multiple Operators. In each Time Zone one of the four Operators can be
programmed.
74. For example, incoming calls are to be routed to the security personnel extension, instead of the Operator when the office is closed
(non-working hours), or certain features in the Class of Service are to be allowed only during working hours, or access to outgoing
long distance calls are to be denied during non-working hours, or the station must play a different greeting message to the callers
during break hours and holidays (non-working hours).
Operator 1 is the default in the Station Basic Feature Template. If you want to assign different stations to
different Operators, you must program a separate Station Basic Feature Template with a different Operator
for each station group.
• Class of Service: Class of Service (COS) defines the set features of the PBX that the extension is to be
allowed access to.
Not all extensions may require the same set of features. Some extensions may require voicemail, while
another group of extensions may need the ability to forward calls to a cell phone, and still others may have
no need to make calls outside the office.
Similarly, certain features may be required during working hours, but not during break or non-working
hours.
It is possible to assign a different Class of Service to different extensions according to their feature
requirements as well as according to the Time Zones.
By default COS group 01 is assigned to the Station Basic Feature Templates for all Time Zones. If you
want to assign a different COS for each Time Zone, you must customize the COS group first and then
assign the number of the COS group in the Template.
Refer the topic “Class of Service (COS)” to know more and for instructions on how to enable or disable a
feature in a COS group.
• Call Budget: This flag is for enabling the Call Budget feature. The Call Budget feature will allot a 'budget'
limit for outgoing calls made by stations on which the template is applied. Refer “Call Budget” for more
details.
• Toll Control Level 0: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed to stations according to the time of the day, during working hours (WH), break hours (BH) and
non-working hours (NH).
For each Time Zone, you may define the calling permission to be allowed to stations by selecting the Type
of Call Privilege.
• Call Privilege: Define the type of calling permission to be allowed to the station during the Working
Hours. The call privilege options are: No Calls, Regional Calls, National Calls International Calls,
Limited Calls.
• Allowed List: If you select 'Limited Calls' as the Call Privilege type, you must define the list of numbers
to be allowed during Working Hours.
• Denied List: If you select 'Limited Calls' as the Call Privilege type, you define the list of numbers to be
denied during the Working Hours.
• Toll Control Level 0 (BH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to a station during Break Hours.
• Allowed List: If you select 'Limited Calls' as the Call Privilege type, you must define the list of numbers
to be allowed during Break Hours.
• Denied List: If you select 'Limited Calls' as the Call Privilege type, you must define the list of numbers
to be denied during the Break Hours.
• Toll Control Level 0 (NH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to a station during Non-Working Hours.
• Call Privilege: Define the type of calling permission to be allowed to the station during the Non-
Working Hours. The call privilege options are: No Calls, Regional Calls, National Calls International
Calls, Limited Calls.
• Allowed List: If you select 'Limited Calls' as the Call Privilege type, you must define the list of numbers
to be allowed during Non-Working Hours.
• Denied List: If you select 'Limited Calls' as the Call Privilege type, you must define the list of numbers
to be denied during the Non-Working Hours.
• Toll Control Level 1: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed a station, regardless of Time Zone. By default Toll Control Level 1 is set to Local Calls.
• Toll Control Level 2: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed a station, regardless of Time Zone. By default Toll Control Level 2 is set to National Calls.
• Toll Control Level 3: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed a station, regardless of Time Zone. By default Toll Control Level 3 is set to No Calls.
• Toll Control - Call Budget Consumed: This Toll Control Level allows you to define the Call Privilege
(calling permission) to be allowed a station, when the Call Budget allotted to the station is consumed. By
default the Toll Control is set to No Calls.
• OG-Trunk Bundle Group (WH): This is the Outgoing Trunk Bundle Group to be allotted to the station for
Working Hours. The station will be allowed to make outgoing calls through the trunks in this group.
• OG-Trunk Bundle Group (BH): This is the Outgoing Trunk Bundle Group to be allotted to the station for
Break Hours.
• OG-Trunk Bundle Group (NH): This is the Outgoing Trunk Bundle Group to be allotted to the station for
Non-Working Hours.
Refer the topic “OG Trunk Bundle Group” for more details.
• Store Outgoing Calls: This flag is used to enable or disable the storage of call details - Station Message
Detail Records - of Outgoing Calls landing on the stations on which the template is applied. Refer the topic
“Station Message Detail Recording-Storage” for more details.
• Click the link 'Station Basic Feature Template' to open the page.
• Change the values of the Station Basic Feature Template parameters as desired.
• Click the link 'SLT Parameters' under SLT Configuration to open the page.
• Go to the SLT software ports to which this Template is to be assigned, for example SLT-003 and 004.
• Click the link 'DKP Parameters' under DKP Configuration to open the page.
• Go to the DKP software ports to which this Template is to be assigned, for example DKP-005 to 008.
• Go to the ISDN Terminal software ports to which this Template is to be assigned, for example ISDN-01.
• Go to the SIP Extensions to which this Template is to be assigned, and enter the Template number.
• Repeat the same steps to customize another template and apply it on the extension ports.
For example, you want to customize Template number 10, by enabling Call Budget feature and select
'Local Calls' as the Toll Control when Call Budget Consumed.
Dial 5502-1-10-06-1 to enable Call Budget feature (number 06) on Template 10.
Dial 5502-1-10-19-1 to select Local Calls (1) as Toll Control when Call Budget Consumed (19) in
Template 10.
Refer the following Table for the Feature Number and Codes for the Station Basic Feature Templates.
For example, to assign Station Basic Feature Template number 10 to the SLT software ports 004 to
010, dial 5503-2-004-010-10.
For example, to assign Station Basic Feature Template 10 to DKP software ports DKP-005 to 010, dial
5504-2-005-010-10
For example, to assign Station Basic Feature Template number 10 to the ISDN Terminal Software
ports 01 to 04, dial 5507-2-01-04-10
• Exit SE mode.
Instead of programming each extension individually, the Station Advanced Feature makes it possible to group
together extensions that are to be assigned the same set of features, prepare a Station Advanced Feature
Template with the common set of features and apply it on these extensions.
The ETERNITY offers 50 such Station Advanced Feature Templates. A Station Advanced Feature Template is
assigned to all the types of stations viz. SLT, DKP, ISDN Terminals, E&M75, T1E1PRI and BRI76.
These templates have commonly used values, but can be customized per the requirement and applied on the
extensions.
• Caller ID Presentation while Transfer: This parameter is related to the CLIP feature. It allows you to
select whether you want the CLI of the ‘Held Party’ or the CLI of the ‘Transferring Party’ to be displayed to
the transfer destination extension while the call is being transferred. Refer the feature description for
“Calling Line Identification and Presentation (CLIP)” to know more.
• DDI IC Routing: This flag is relevant only if you are using DDI IC Routing. As this flag is enabled by
default, hence all the stations configured in the IC Reference Table will behave as DDI stations. If you do
not want any of these stations to function as a DDI station, disable this flag in the feature template applied
on that station. By default, the flag is enabled.
Refer the topics “Direct Dialing-In (DDI)”, “DDI Routing Table” and “IC Reference Table” to know more.
• Send DDI Number as CLI?: This flag is relevant only if the station is programmed as a DDI Station (the
DDI IC Routing is enabled). You can choose whether to the Calling Line Identification (CLI) of the DDI
station should be sent for outgoing calls made from that station. By default the flag is enabled.
• Internal Calls Storage: This parameter is related to the storage of Station Message Detail Records of
internal calls, made between stations of the ETERNITY. You can select the type of internal calls to be
stored by the system: i) Calls made from the station, ii) Calls made to the station, iii) Calls made from as
well as calls made to the station, and iv) all types of calls, and v) no internal calls. By default, all types of
internal calls on the station are stored. Select desired type of internal call storage.
• Walk-Out Mode: This parameter is related to the feature Walk-In Class of Service. ETERNITY offers two
types of Walk-In: i) One-Call per Walk-In, whereby the user is automatically logged out after a call. ii) Walk-
In until Logout, whereby the user remains logged on until s/he manually walks out or a second user walks
into the same station.
You must select the Walk-Out mode for the station. If One-Call per Walk-In is to be supported on the
station, select 'One Call' as Walk-Out mode.
If Walk-In until Logout is to be supported on the station, select 'Multiple Calls' as the Walk-Out mode.
• CDC Table: This parameter is to be programmed if you have enabled the “Call Duration Control (CDC)”
feature on the station. The system will check the Call Duration Control (CDC) Table applied to station to
implement this feature on the station. So, you must first program the CDC Table and enter the number of
the CDC Table you have programmed in this field.
You can program 8 different CDC tables. By default, CDC Table No. 1 is assigned to all stations. If CDC is
to be applied on stations of the ETERNITY, simply program the default CDC Table No. 1.
To do this, click the link CDC Table to open the page. Program the CDC Table parameters and 'Submit' to
save your settings. Now return to the Station Advanced Feature Template and enter the number of the
CDC Table you programmed in the CDC Table field of the template.
Refer the feature description “Call Duration Control (CDC)” to know more and for instructions on creating
CDC Tables.
• Forced Account Code: This flag used to enable or disable the feature Forced Account Code on the SLT/
DKP/ISDN Terminal extensions. When this flag is enabled, the system will allow the extension user to dial
• Department Billing Group: This parameter enables you to know the total cost of the calls made by a
particular group of stations. This parameter is used as a one of the filters for printing SMDR Reports
namely, “Print outgoing calls department group wise”. To be able to use this filter, you must assign the
station to a Department Bill Group. You can create as many as 99 different Department Bill Groups. Enter
the number of the Bill Group you want to assign the stations to in this field.
• Floor Service Group: This parameter is related to the Floor Service feature. Floor Service can be floor-
wise or centralized. Floor Service requires you to program Routing Groups as landing destinations for
extension calls.
Program the Floor Service (Routing) Group first and enter this Floor Service (Routing) Group number in
this field. There are 96 different Routing Groups to be programmed as Floor Service Groups. By default,
no Routing Group is assigned to Floor Service in the Template ('00').
To know more about this feature, refer the feature description for “Floor Service”.
Calls from the station will land on the Floor Service (Routing) Group you have assigned in this field.
• Alarm Notification Type: This parameter is related to the Alarm feature of the ETERNITY. You can select
any from the following options for Alarm Notification to station users for Alarm calls:
• Voice Message: Extension users will be played a message recorded in the Voice Module, when they
answer the alarm call. By default, this option is selected.
• Music-on-Hold: Extension users will be played music-on-hold when they answer the alarm call.
• External Music Source: Extension users will be connected to live music when they answer the alarm
call. If you select this option, make sure you have connected a live music source to the Analog Input
Port of the ETERNITY.
• Routing Group: Extension users will be connected to the stations programmed in the 'Alarm
Notification Group'. For this you must have programmed a Routing Group.
If you select this option, you can also connect external Messaging Devices to play real-time updated
information like date, time, greetings, weather information, specific event announcements, etc, when
then extension users answer the alarm calls. The messaging devices must be connected to any of the
SLT ports and included in the Routing Group for alarm notification.
• Alarm Notification Routing Group: Program this parameter if you have selected 'Routing Group' as the
Alarm Notification Type. Enter the number of the Routing Group you have programmed for Alarm Calls.
By default Routing Group 32 is assigned as the Alarm Notification Routing Group. If the same Routing
Group is to be assigned to all stations, click the link 'Alarm Notification Routing Group' to open the Routing
Groups page. Select members (stations) in this routing group. Save your changes by clicking 'Submit'
button.
You can program a different Routing Group repeating these steps. Make sure to enter the number of the
Routing Group you programmed in this field.
If the station is to be configured as the master station, configure the following parameters:
• Master Port Type: Select the port type of the station, whether SLT or DKP. If you are not using any
Virtual Station, select 'Null'.
• Port Number: Enter the software port number of the station designated as the master station in this
field.
• Help Desk: Enable this flag if you want to define the station as a “Help Desk”. When this flag is enabled,
Auto Call Back will be automatically set whenever this station is found busy.
• GPAX - Charge Internal Calls: This parameter is related to the GPAX application. If the station is
programmed as a GPAX user, enable this flag for billing internal calls made by the station. When this flag is
enabled, the system will record all calls made from the station in the Station Message Detail Record-
Outgoing buffer for “GPAX Billing”. If the flag is disabled the calls will not be billed and will be recorded in
the Station Message Detail Record - Internal buffer as an internal call.
• Call Taping: If you want to use the “Call Taping” feature on the station, you must program the related
parameters described below.
• Tape Calls without CLI?: Enable this flag if you want incoming calls without CLI to be taped. By
default, the flag is disabled. The system will not tape incoming calls without CLI.
• Number List-Incoming Calls: Assign a Number List containing numbers of Incoming Calls that must
be taped. You must first program the Number List. By default, Number List 09 is assigned.
• Number List-Outgoing Calls: Assign a Number List containing numbers of Outgoing Calls that must
be taped. You must first program the Number List. By default, Number List 10 is assigned also for
outgoing calls.
If Number list 10 is already used for another application, prepare a different number list and assign it to
the template.
• Call Taping for Internal Calls: Enable this flag if you want to allow Call Taping of internal calls made
and received by the station.
b. If you want to define stations of the PBXs networked over E&M lines using MFCR2 Signaling as either
'Priority Subscriber' or 'Ordinary Subscriber'.
• Allow External Call Forward for: This parameter defines the types of calls for which the External Call
Forwarding is to be applied. This parameter is relevant for the features “Call Forward” and “Mobility
Extension”. You may select from the following options:
• Internal Calls Only
• Trunk Calls Only
• Internal + Trunk Calls.
• Click the link 'Station Advanced Feature Template' to open the page.
• Select a Template number you wish to customize, for example Template 02.
• Change the values of the Station Advanced Feature Template parameters as desired.
• Click the link 'SLT Parameters' under SLT Configuration to open the page.
• Go to the SLT software ports to which this Template is to be assigned, for example SLT-003 and 004.
• Enter the number of the Template you customized, 02, in the field 'Station Advanced Feature Template' of
each of these SLT ports.
• Click the link 'DKP Parameters' under DKP Configuration to open the page.
• Go to the ISDN Terminal software ports to which this Template is to be assigned, for example ISDN-01.
• Repeat the same steps to customize another template and apply it on other extension ports.
For example, you want to customize Template number 02, by changing the default settings of the
following features:
• Call Forward No Reply Timer to set to 45 seconds, dial 5602-1-02-02-045 (Template 02, Feature
Number 02 for No Reply Timer, Code 045 for 45 seconds).
Refer the following Table for the Feature Number and Codes for the Station Advanced Feature Templates.
For example, to assign Station Advanced Feature Template 02 to DKP software ports DKP-005 to 010,
dial 5604-2-005-010-02
For example, to assign Station Advanced Feature Template number 02 to the ISDN Terminal Software
ports 01 to 04, dial 5607-2-01-04-02
• Exit SE mode.
The number of SLT extensions available to you for configuration depends on the number of SLT ports supported by
your model of ETERNITY and the number of SLT ports you have specified on the “System Pre-requisites” page.
If you have enabled 'On-Site Configuration', the system will provide you only those ports that are actually present in
the system for configuration.
Configure SLT port parameters using Jeeves or by dialing commands from a Telephone.
• Configure the following parameters for each SLT port on this page:
• SLT Port No.: This non-editable field is the number of the software port of the SLT port.
• Hardware - Slot - Port: 'Slot' is the number of the Universal Slot in which the SLT Card is inserted.
'Port' is the number of the SLT hardware port on which the telephone instrument is connected.
For example: if you have inserted the SLT8 Card in Slot number 02 and SLT16 Card in Slot number 03
of ETERNITY ME16S, the system will assign the hardware slot 02 and port numbers 01-08 to the SLT
Software Ports from 001 to 008 respectively. The system will assign Slot 03 and port numbers 09-24 to
the SLT Software Ports 009 to 024. Refer the topic “Software Port and Hardware ID” to know more.
However, if required, you may change the Hardware Slot and Port assigned to the SLT software port. In
which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
• Access Code: Assign Station Access Codes to the SLT Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be a combination of 1, 2, 3 or 4 digits, which are dialed
to call the SLT port to which they are assigned.
All SLT ports are assigned the following Station Access Codes as default.
001 2001
002 2002
003 2003
: :
: :
512 2512
You may either apply the default Station Access Codes to the SLT ports or assign them according to
your requirement and preferences.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics “Access Codes” and “Conflict Dialing” to know
more.
• Name: Assign a 'Name' to the SLT port. The name may be of the person who will use the SLT or the
name of a department. This name will be displayed on the LCD of the remote user's phone, if it is
equipped with Caller ID.
• SLT Hardware Template: Assign an SLT Hardware Template to the SLT port. An SLT Hardware
Template is a set of features that define the behavior of the SLT hardware port. The SLT Hardware
Template allows you to configure according to user requirements a common set of features for all SLT
Hardware Ports, like Caller ID Presentation (DTMF, FSK), Digit Pad Count, Ring Type, AC Impedance,
Answer Signaling type, Speech Transmit and Receive Gains, Open Loop Disconnect, Loop Current,
and external Voice Mail and Fax connectivity.
If the default SLT Hardware Template 01 fulfills the feature requirements and if the same features are to
be allowed to all SLTs, retain Template 01.
If different sets of hardware features are to be allowed to different SLTs, then prepare separate SLT
Hardware Templates and apply them on the ports. To do this,
• Customize Template number 02 and click Submit at the bottom of the page.
• Enter the number of the Template you customized, template 02, in the SLT Hardware Template
field of the SLT Ports on which you want to apply this template.
• Repeat the same steps to customize and assign a different SLT Hardware Template to another SLT
port.
Also, refer the topic “SLT Hardware Template” to know more about customizing the templates and
applying on the SLT ports.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the SLT. A Station
Basic Feature Template includes a set of features that completely define the behaviour of the extension
port, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call Budget,
and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all stations of the system that includes
DKP ports, ISDN Terminals, SIP extensions, and E&M Lines with Station as Orientation Type.
Check if the default template fulfills the feature requirements (like “Class of Service (COS)”, “Toll
Control”, “OG Trunk Bundle Group”, etc.) of the SLT.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all SLTs, retain Template 01.
If different sets of features are to be allowed to different SLTs, then prepare separate Station Basic
Feature Templates and apply them on the ports. To do this,
• Click the link Station Basic Feature Template to open the page.
• Customize Template number 10 and click Submit at the bottom of the page.
• Enter the number of the Template you customized, Template 10 in the Station Basic Feature
Template field of the SLT Port, for instance SLT-003, on which you want to apply this template.
• Repeat the same steps to customize and assign a different Template to another SLT port.
Also, refer the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on the ports.
• Station Advanced Feature Template: Assign a Station Advanced Feature Template to the SLT. The
Advanced Feature Template consists of a set of less commonly used features like Message Wait
Notification Type, Alarm Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming
Call Routing, Storage of Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all stations of the ETERNITY, which
includes DKP Ports, ISDN Terminals, SIP Extensions, and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the SLT Ports by clicking the link
Station Advanced Feature Template.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all SLT ports, retain Template 01.
If different sets of features are to be allowed to different SLT Ports, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
• Click the Station Advanced Feature Template link to open the page.
• Enter the number of the Template you customized, template 02, in the Station Advanced Feature
Template field of the SLT Ports on which you want to apply this template.
• Repeat the same steps to customize and assign a different Template to another SLT port.
Also refer the topic “Station Advanced Feature Template” for instructions on customizing these
templates and applying them on the ports.
• Call Pick-Up Group: Configure this parameter if you want to assign the SLT to a particular “Call Pick
Up” group.
For this to work, both extensions, the ringing extension and the station picking up the call, must be in
the same Call Pick Up Group. Refer “Call Pick Up” for instructions on how to create groups. You can
create as many as 99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SLT in this field.
• Station Type: This parameter is relevant only for the Hotel Application of the ETERNITY. Extensions
are identified as 'Administrator' or 'Guest' extensions according to the intended user of the SLT. When
the Station Type is selected, the system will automatically assign the Guest and Administrator (Hotel
Staff) features to the SLT. To know more, refer the ETERNITY Hospitality System Manual.
Advanced Configuration
For extension users who need to be provided features like Personal Directory or assigned a Priority, you may click
the Advance button at the bottom of the page and program the following parameters:
• Personal Directory: Enter the number of the Personal Directory that you want to assign to this SLT. A
Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (“OG Trunk Bundle Group”). The Personal Directory is
necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension phone with higher priority calls an
extension with lower priority, a triple ring is placed on the called station. To know more, read the feature
description “Priority”.
By default, the Priority of all SLT ports is set to '5'. So, decide what Priority Level you will assign to each of
the SLTs and set the desired level for each port.
• If you have completed configuring SLT parameters, click Submit at the bottom of the page to save your
settings.
It is possible to default all the parameters by clicking the Default button. You can also restore default
values of the parameters of a single SLT by clicking the Default One button and specifying the SLT you
want to set to default.
To de-assign the hardware slot and the hardware port of an SLT port, dial:
• 1101-SLT-00-00
To define the Station Type for an SLT Port, Refer ETERNITY Hospitality System Manual.
• Exit SE mode.
The number of DKP extensions available to you for configuration depends on the number of DKP ports supported
by your model of ETERNITY and the number of DKP ports you have specified on the “System Pre-requisites” page.
If you have enabled 'On-Site Configuration', the system will provide you only those ports that are actually present in
the system for configuration.
Configure DKP port parameters using Jeeves or by dialing commands from a Telephone.
The DKP Ports appear in tabs, with eight DKP Ports in each tab, 001-008, 009-016, 017-024, and so forth.
• DKP - Slot - Port: 'Slot' is the number of the Universal Slot in which the DKP Card is inserted. 'Port' is
the number of the DKP hardware port on which the proprietary DKP EON is connected.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the DKP software ports.
However, if required, you may change the Hardware Slot and Port assigned to the DKP software port.
In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
If your ETERNITY is ME10S or ME16S, the system will detect and assign the first four Software Ports to
the four DKP Ports located on the Switch Card.
• Access Code: Assign Station Access Codes to the DKP Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be a maximum of 6 digits, which are dialed to call the
DKP port to which they are assigned.
All DKP ports are assigned the following Station Access Codes as default.
001 3001
002 3002
003 3003
: :
: :
128 3128
You may either apply the default Station Access Codes to the DKP ports or assign them according to
your requirement and preferences.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics “Access Codes” and “Conflict Dialing” to know
more.
• Name: Assign a 'Name' to the DKP port. The name may be of the person who will use the DKP or the
name of a department. This name will be displayed on the LCD of the DKP of the user and on other
extension user's phones, provided these are also a model or EON or are equipped with Caller ID.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the DKP.
A Station Basic Feature Template includes a set of features that completely define the behaviour of the
Station, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call
Budget, and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all stations of the system that includes
DKP ports as well as DKP ports, ISDN Terminals, E&M Lines with Station as Orientation Type, and SIP
Extensions.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all DKPs, retain Template 01.
If different sets of features are to be allowed to different DKPs, then prepare separate Station Basic
Feature Templates and apply them on the ports. To do this,
• Click the link “Station Basic Feature Template” to open the page.
• Customize Template number 11 and click 'Submit' at the bottom of the page.
• Enter the number of the Template you customized, Template 11, in the 'Station Basic Feature
Template' field of the DKP Port, for instance DKP-005, on which you want to apply this template. If
you want to apply this template to other ports too, like DKP-006, 007, and 008, assign the Template
11 to all these ports.
• Repeat the same steps to customize and assign a different Template to another DKP port.
Also, refer the topic Station Basic Feature Templates to know more about customizing the templates
and applying on the ports.
• Station Advanced Feature Template: Assign a Station Advanced Feature Template to the DKP. The
Advanced Feature Template consists of a set of less commonly used features like Message Wait
Notification Type, Alarm Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming
Call Routing, Storage of Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all stations of the ETERNITY, which
includes DKP Ports, ISDN Terminals, E&M Lines configured as Stations, and SIP Extensions.
Check if this default template fulfills the feature requirements of the DKP Ports by clicking the link
'Station Advanced Feature Template'.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all DKP ports, retain Template 01.
If different sets of features are to be allowed to different DKP Ports, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
• Click the 'Station Advanced Feature Template' link to open the page.
• Select a Template number, for example 02.
• Customize Template number 02.
• Enter the number of the Template you customized, Template 02 in the 'Station Advanced Feature
Template' field of the DKP Port, for instance DKP-001, on which you want to apply this template. If
you want to apply this template to other ports too, like DKP-002, 003, and 004, assign the Template
04 to all these ports.
• Repeat the same steps to customize and assign a different Template to another DKP port.
Also refer the topic “Station Advanced Feature Template” for instructions on customizing these
templates and applying them on the ports.
• Call Capacity: Call Capacity is the number of Call Appearances (also referred to as 'call loops')
assigned to a (DKP) station. It is the ability of a Station to handle multiple calls simultaneously. A Call
Appearance allows a station user to attend to more than one calling party at a time.
A minimum of two Call Appearances must be assigned to a DKP Station - Operator station or Executive
station - so that the Station user can put one party on hold while talking to another. A third Call
Appearance allows the station user to put two calls on hold, make/attend a third call and toggle
between three calls.
The higher the call capacity (the more the number of Call Appearances assigned to a station), the more
the number of calls the Station user can handle.
The ETERNITY supports a maximum of 10 Call Appearances as Call Capacity of the DKP extensions.
DKP stations for Executives are usually assigned 2 Call Appearances, while the Operator Station is
assigned 6 Call Appearances to handle 6 calls simultaneously. The default call capacity of the DKP
ports is 02.
Now, select the number of Call Appearances you wish to assign to the DKP port in the column, 'Call
Capacity'.
• Key Map: EON is designed to function as Operator, Executive, Hotel Attendant, and Hotel Guest
extensions, providing default key settings (key maps) for all these functions. All you need to do is
assign a Key Map Template according to the intended user of the DKP.
For example, if the DKP is to be used by the Operator, select ‘Operator's Template’. The DKP will be
assigned the key template with the special features required by Operators, such as more DSS keys for
Trunk Access and Call Appearances, a Call Release Key, etc.
Similarly, if the user of the DKP is a Hotel Attendant, select 'Hotel Attendant's Template'. The key map
with the specific Front Desk User features such as Check-In, Check-Out, Guest In/Out, Change Room
Clean Status, Room Shift, will be automatically assigned to the DKP.
You can also customize the key map of the DKP, by selecting the option ‘Personalized’ and assign
functions to keys as per your requirement.
• Call Pick-Up Group: Program this parameter if you want to assign the DKP to a particular “Call Pick
Up” group.
Call Pick Up allows the DKP station user to 'pick up' (answer) calls ringing on any other station, by
dialing a feature code, without physically going to the ringing station. For this to work, both the ringing
station and the station picking up the call must be in the same 'Call Pick Up Group'. Refer “Call Pick
Up” for instructions on how to create groups. You can create as many as 99 groups numbered from 01
to 99.
Enter the number of the Call Pick-Up Group you created for this DKP in this field.
• Station Type: This parameter is relevant only for the Hotel Application of the ETERNITY. Extensions
are identified as 'Administrator' or 'Guest' extensions according to the intended user of the DKP. When
the Station Type is selected, the system will automatically assign the Guest and Administrator (Hotel
Staff) features to the DKP. To know more, refer the ETERNITY Hospitality System Manual.
For instructions on configuring EON Keys, DSS1 Keys and DSS2 Keys, see the topic “DSS Keys Programming”
Advanced Configuration
The above listed parameters fulfill the basic DKP station port configuration requirements of most users. However, it
is anticipated that some users may need to configure the more advanced features like Personal Directory,
Language Selection, DKP Settings - Backlight Brightness and Contrast, Headset/Handset/Speaker Volume levels,
Select a Ringer Tune, attach Direct Station Selection Consoles, etc. For such users, you may click the 'Advanced'
button.
• Personal Directory: Enter the number of the “Personal Directory” that you want to assign to the DKP. A
Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (“OG Trunk Bundle Group”). The Personal Directory is
necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
To be able to assign a Personal Directory to a DKP you must first program it. Refer the topic “Abbreviated
Dialing” for instructions on programming the Personal Directory.
• Priority: Select a Priority Level for the DKP from 1, 2, 3... to 9, with '1' being lowest Priority and '9' being
highest Priority. Whenever a station (DKP) with higher priority calls a station with lower priority, a triple ring
is placed on the called station. To know more, read the feature description “Priority”.
• CO CLIP Pattern: This parameter allows you to select the type of Calling Line Presentation on the DKP for
incoming calls from trunks. You can select any of the below options:
By default, Number + Name is selected as the CO CLIP Pattern for all DKPs.
• Language: ETERNITY provides language support for English, French, German, Spanish, Portuguese,
and Italian. When you select any of these languages, all the command strings and prompts will appear in
the selected language. By default English is selected.
• Ringer Mode: You can select a Ringer mode for each DKP from the four options:
• Ring immediate (it rings immediately as a fresh calls lands on the DKP).
• Ring if idle (rings only if the DKP is idle).
• Ring after a delay (if the call is still not answered).
• Ring Off (silent mode).
By default the Ringer Mode is set to 'Ring Immediate'. Change the Ringer Mode on the DKP as per the
requirement and preferences of the DKP users.
• Ring Delay Timer: The Ring Delay Timer is the time in seconds the ETERNITY will wait to ring on
receiving a call. This Timer needs to be set only if you have selected 'Ring after a delay' as the Ringer
Mode for the DKP in the previous parameter.
By default, the Ringer Delay Timer is set to 10 seconds. You may change the Ring Delay Timer according
to the preferences and requirements of the DKP user.
• Acknowledgement Timer: This Timer is to be programmed to enable the Ringer Auto Acknowledge
mode. This mode determines when to stop the ring on the DKP. There are two options for ringer auto
acknowledge:
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
• Play Ring ON: With this parameter, you can assign the Ring Destination for the DKP; you can choose
whether the Ring should be played on the Speakerphone or Headset of the DKP. Default: Speakerphone is
selected.
When you select the Headset as the destination, ensure that you have enabled “Headset Connectivity” flag
and have connected a Headset to the DKP.
The speech path of both, the Headset and the Handset is common. If the Headset is not connected and
you have selected the Headset as the ring destination, the ring will be played on the speaker of the
Handset.
• Ring Tune: You can select from different ringer tunes for each DKP according to your preferences and
requirement. By default, Ringer Tune is set to 1.
• Ring Volume: You can select from different ringer volumes for each DKP according to your preferences
and requirement. By default, Ringer Volume is set to 5.
• Handset MIC Transmit (Tx) Volume Level: This parameter is used for increasing or decreasing the
volume of outgoing speech (Transmit Gain) on the Handset of the DKP. Select the desired Handset Tx
Volume Level from 0 to 9. By default, Handset Tx Volume Level is 5.
• Handset Rx Volume Level: This parameter is used for increasing or decreasing the volume of incoming
speech (Receive Gain) on the Handset of the DKP. Select the desired Handset Rx Volume Level from 0 to
9. By default, Handset Rx Volume Level is 4.
• Headset Transmit Volume Level: This parameter is used for increasing or decreasing the volume of
outgoing speech (Transmit Gain) on the Headset port of the DKP. Select the desired Headset Tx Volume
Level from 0 to 9. By default, Headset Tx Volume Level is 4.
• Headset Rx Volume Level: This parameter is used for increasing or decreasing the volume of incoming
speech (Receive Gain) on the Headset port of the DKP. Select the desired Headset Rx Volume Level from
0 to 9. By default, Headset Rx Volume Level is 4.
• Hands-free Tx Volume Level: With this parameter you may change the Volume level of Transmit Gain of
the Speaker phone MIC volume from 0 to 9, as per your preference. This parameter is to be used for
increasing or decreasing the volume levels of outgoing speech on the Speaker of the DKP. By default,
Hands-free Tx volume level is 4.
• Hands-free Rx Volume Level: With this parameter you may change the Volume level of Receive Gain of
the Speaker phone MIC volume from 0 to 9, as per your preference. This parameter is to be used for
increasing or decreasing the volume levels of incoming speech on the Speaker of the DKP. By default,
Hands-free Rx volume level is 4.
• Key Click Volume Level: You may change the Key Click Volume (Key DTMF Side tone) of the DKP. Key
Click Volume is the tone you hear as you press the dial pad keys of EON. Select the desired volume level
from 0 to 9. By default, the volume level is set to 5.
• DTMF Transmit Level: You can select the desired Transmit Level from 0 to 9 for DTMF generation from
the DKP. By default, the DTMF Transmit Level is set to 5.
• Headset Connected?: Enable this parameter by selecting the checkbox if you want to use a Headset with
the DKP.
Make sure that you have also connected a Headset to the DKP.
• Auto Answer: Enable this parameter by selecting the checkbox if you want to set the “Auto Answer”
feature on the DKP.
When this feature is set, the DKP goes OFF-Hook automatically after a preset period of time, without the
user having to pick up the handset or press the speaker or headset key.
• Auto Answer Timer (sec): This parameter is to be programmed if you have enabled the “Auto Answer”
feature on the DKP.
When the Auto Answer feature is enabled, the Auto Answer Timer must be programmed. This timer
defines the time in seconds that the DKP should wait before going OFF-Hook. The range of this timer is 1
to 9 seconds. By default, the Auto Answer Timer is set to 1 second.
If you set the Auto Answer Timer to 0, Auto Answer will be disabled.
• LCD Backlight Level: You can change the LCD Backlight Brightness of EON. The intensity of the
backlight brightness increases from 0 to 4, where '0' will cause the backlight to be turned OFF. '1' signifies
minimum intensity, '4' signifies maximum intensity. Select any of the levels from 1to 4 from the list.
• LCD Backlight OFF Timer: The backlight of the LCD display of EON can be kept switched ON
continuously, or can be set to switch OFF automatically after a predefined period of time, known as the
Backlight OFF Timer. The range of the Backlight OFF Timer is 000-999 seconds. By default it is set to
switch OFF after 060 seconds.
• LCD Contrast Level: The EON offers 4-level contrast control for its LCD display. Level 1 signifies
minimum and level 4 is the maximum. The contrast increases in steps of 1 to 4. By default the contrast is
set to level 3. You may adjust it to the level comfortable to you. Select a level from 1-4.
• DSS1 - Slot/Port: This parameter is to be configured only if you have attached a Direct Station Selection
Console with the DKP.
You can attach two “Direct Station Selection Console” to each DKP to increase the number of DSS keys
for Direct Station Calling. You may attach any two DSS Consoles of the same model or of two different
models to the DKP.
DSS Consoles are connected to the DKP Cards just like digital key phones (refer the topic “Installing DSS
Consoles” in the installation instructions for your model of ETERNITY.
If you have attached a single DSS Console to a DKP, you must enter the following information in the 'DSS1
H/w-Slot to Port' column:
• the number of the Port on the card, on which the DSS Console is connected.
• DSS2- Slot/Port: To configure a second DSS Console attached to the DKP, enter the same information as
above here, that is, the number of the Universal Slot in which the DKP card to which the DSS Console is
located, and the number of the Port on the card, on which the DSS Console is connected.
The DSS Console you want to attach to a particular DKP must not necessarily be located on the same card
as the DKP to which it is attached.
For example, you want to attach two DSS Consoles to DKP Port Number 001 in Slot number 18. The first
DSS Console is connected to DKP Port Number 01 on the card which is occupying Slot number 02. The
second DSS Console is connected to the DKP Port Number 02 on the same card. Now, to attach the DSS
Consoles to the DKP connected on Port Number 001, enter the hardware ID and port number as follows:
Enter 02 and 01 as the hardware ID and port number in the column DSS1. Similarly, enter 02 and 02 as
the hardware ID and port number in the column DSS2 of the DKP port.
001 17 01 02 01 02 02
002 17 02
003 17 03
004 17 04
005 17 05
006 17 06
007 17 07
008 17 08
009 02 01
010 02 02
• If you have completed configuration of all the above listed DKP Parameters, click 'Submit' to save your
changes.
Two additional DKP Parameters you can configure only by dialing the related SE Commands from a telephone are:
• Rx Gain at SIP Trunk/Extension (dB): This parameter allows you to increase the incoming speech
volume level of calls from SIP trunks/Extensions to Mobile Ports. By default, Rx Gain is set to 0dB.
• Tx Gain at SIP Trunk/Extension (dB): This parameter allows you to increase the outgoing speech
volume level of calls from SIP trunks/Extensions to Mobile Ports. By default, Tx Gain is set to 6dB.
To clear the hardware Slot and Port assigned to the DKP software port, dial:
• 1102-DKP-00-00
To select a Station Type for a DKP Port, refer the ETERNITY Hospitality System Manual.
• For Advanced Configuration of the DKP Ports, use the following commands:
To select Destination for 'Play Ring ON' for a DKP Port, dial:
• 1220-1-DKP-Ring Destination to select Play Ring ON destination for a single DKP port.
• 1220-2-DKP-DKP-Ring Destination to select the same Play Ring ON destination for a range of DKP
ports.
• 1220-*-Ring Destination to select the same Play Ring ON destination for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Ring Destination is
1 for Play Ring on Speaker Phone
2 for Play Ring on Headset
Setting volume level to '0' will cause the Handset MIC volume to be turned OFF.
Setting volume level to '0' will cause the Handset Speaker volume to be turned OFF.
To set Headset Transmit (Tx) Transmit Volume Level for a DKP, dial:
• 1222-1-DKP-Headset MIC Volume Level to set receive volume for a single DKP port.
• 1222-2-DKP-DKP- Headset MIC Volume Level to set the same receive volume level for a range of
DKP ports.
• 1222-*- Headset MIC Volume Level to set the same receive volume level for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Headset MIC Volume Level is from 0 to 9. Default: 5.
• Exit SE mode.
The DKP parameters described above are set by the System Engineer.
• In addition to these parameters, the users of the DKP can change several phone settings according
their preferences and requirement. These are referred to as DKP Personal Settings and include:
• Ringer Volume
• Ringer Tune
• Ringer Mode
• Ringer Acknowledge Mode
• Speech Volume (Transmit/Receive)
• Time Zone (Working Hour/Break Hour/Non-working Hour)
• User Status (Present/Absent)
• Keypad Security (Lock/Open)
• Call Answer Type - Manual/Auto
• Headset/Handset Connectivity option
To be able change the DKP personal settings, the DKP users must access and navigate the phone menu.
Refer “Digital Key Phone-Operation”.
Key Templates
EON, the proprietary digital key phone (DKP) of Matrix and SETU VP248, the Extended IP Phone of Matrix, can be
the extension of the Operators and Executives in an enterprise, and in hotels, it can be the extension of the Front
Desk Attendants and Guests.
Each of these groups of users may require a different set of features on their phones. For example, when EON/
SETU is an Operator's extension, for efficient call management, more DSS keys may be required for Trunk Access,
Call Appearances, Call Release, Direct Station (DKP/SLT) Calling, than for accessing features.
When EON/SETU is an Executive's extension, more DSS keys may be required for single-touch access to
features, and fewer keys for Trunk Access and Direct Station Calling.
Similarly, when EON/SETU is a Hotel Attendant's extension, keys are required for specific hotel functions such as
Check-In/Check-Out, Changing Room Clean Status, Room Shift, etc. But, a different set of keys with special
functions are required when EON/SETU is provided as a guest extension, because guests are allowed only a
limited number of features of the ETERNITY, such as calling the Front Desk/Floor Service, setting Do Not Disturb,
Wake-up Calls, Call Forward, and checking Voice Mail.
Given the varying requirements of these groups of extension users, ETERNITY provides programmable Templates
of Key Maps for the Operator, Executive, Hotel Attendant, and Hotel Guest.
The default Key Maps in these Templates can be customized to match the requirement of the intended user group.
The customized Template is assigned to the DKPs/Extended IP Phones. For example, you may customize the Key
Template for the Operator and assign it to the Operator DKPs. Likewise, you may customize the Key Template for
Executive and assign it to the Executive extensions.
The default Key Maps vary according to the model of the DKP/Extended IP Phone in use, as illustrated below.
The keys in the Key Templates are numbered only for the purpose of locating the keys when programming.
Key numbers do not appear on the key labels on the phone body.
1 2 3 1 2 3 1 2 3
4 5 6 4 5 6 4 5 6
7 8 9 7 8 9 7 8 9
* 0 # * 0 # * 0 #
25 25 25
20 21 22 26 27 28 20 21 22 26 27 28 20 21 22 26 27 28
CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold
01 09 2 01 09 2 01 09 2
1 abc 3 def 1 abc 3 def 1 abc 3 def
02 10 02 10 02 10
4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno
03 11 03 11 03 11
07 CA 2 15 07 CA 2 15 07 CA 2 15
08 CA 1 16 29 08 CA 1 16 29 08 CA 1 16 29
20 21 22 26 27 28 20 21 22 26 27 28 20 21 22 26 27 28
CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold
Local Local
01 09 01 09 01 09
Menu 1 2 abc 3 def Menu 1 2 abc 3 def 1 2 abc 3 def
02 10 02 10 02 10
4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno
03 11 03 11 03 11
07 CA 2 15 07 CA 2 15 07 CA 2 15
08 CA 1 16 29 08 CA 1 16 29 08 CA 1 16 29
By using Key Templates you can prepare and assign common key maps to all or as many DKPs and Extended IP
Phones as you want, at one go.
ETERNITY also offers the flexibility to personalize the Key Maps of each DKP/Extended IP Phone, instead of using
the Key Templates. For example, if you have assigned a common Executive Key Template to 12 DKPs, but you
want to reassign some of the keys on two of these DKPs, ETERNITY allows you to selectively personalize the key
maps of these two DKPs.
• List the features/facilities that you want to change in each of the existing (default) Key Templates of
Operator, Executive, Hotel Attendant, and Guest.
• For each template, decide the keys that will be reassigned the features you listed.
• You may use the key templates printed above to decide the position of keys.
• For each template that you customize, list down the DKPs which will be assigned the template,
along with their Software port numbers and their corresponding Hardware Slot and Port Offset.
Similarly list Extended IP Phones along with their Software port numbers (SIP Extension numbers)
and VoIP Port numbers.
• Similarly, list the DKPs and the Extended IP Phones which are to be assigned personalized Key
Maps, along with their Software port numbers and their corresponding Hardware Slot and Port
Offset.
Illustrated below is an example of a customized Operator's Template for the EON48 model:
You can customize the key template and assign it to DKPs using Jeeves or a Telephone.
EON48 and SETU VP248 have 12 Touch-sense feature keys. While you can reassign the features on
these keys, you cannot re-label the keys. Avoid reassigning features on touch-sense keys.
• The links to the default key templates for Operator, Executive, Hotel Attendant and Guests for EONSOFT,
each model of EON and the Matrix Extended IP Phone appear on your screen.
• Select the Key Template you want to customize according to the model of EON/Extended IP Phone in use,
by clicking the respective link. In case you have more than one model of EON, you may customize the
desired Key Template for each model.
Let us attempt to program the sample Operator template we customized earlier for EON48.
• Click the 'Operator' template link of EON48.The Operator Key template for EON48 will appear on your
screen.
DKP2 DND-Override
• To assign Trunk to Trunk Transfer, click the DKP4 key. A new window opens.
All features that can be assigned to keys will appear in the 'Select Offset' list.
• The 'Trunk to Trunk Call Transfer' feature will appear in abbreviated form as ‘T-T Xfer’ on the key label.
• As Remote DND is an SA Command, select the option 'SA Command' in the Select Function Type list.
• Click OK. The window closes. The Remote DND feature appears in abbreviated form as ‘R-DND’ on the
key label.
• Repeat these steps to reassign other keys, selecting the appropriate Function Type and the Offset for each
feature/function.
To assign direct access to Mobile Trunk 1, select “Mobile” as function type and '01' as Offset.
To assign direct access to SIP Trunk 1, select “SIP” as function type and '01' as Offset.
To assign direct access to BRI Trunk 1, select “BRI” as function type and ‘01’ as Offset. When you select
BRI as function type, you will be asked to ‘Select Channel’. Since you do not want to assign direct access
to any particular BRI channel of this trunk, retain the option ‘All Channels” for ‘Select Channel’.
When you select T1E1, as function type, you will be given the ‘Select Channel’ option. Since you want to
assign direct access to Channel 2 of T1E1 Trunk 1, select the option ‘Channel 2’ in the ‘Select Channel’
list.
Always click 'OK' when you select a Function Type and Offset.
• When you have completed assigning functions to keys, click 'Submit' at the bottom of the page to save
your settings.
To take the above example further, the Operator key template customized for EON48 is to be assigned to
the DKPs 001 and 002. To do this,
• Scroll with the horizontal bar to reach the parameter 'Key Map' of this DKP 001.
• Click 'Submit'.
• Also refer the topic “Configuring DKP Parameters using Jeeves” for instructions on assigning the
Key Map in the DKP Parameters.
b. Function - the function that the key should perform, that is, as direct access to a feature or an SLT or DKP
station, a trunk type, an SA command, etc.
Follow the numbering of keys on the default Key Maps illustrated earlier in this topic.
To assign the same feature to the same key number in a range of templates, dial:
To assign the same feature to the same key number in all the templates, dial:
Where,
Key Template is
1 for Operator
2 for Executive
3 for Hotel Attendant
4 for Guest
Terminal Type is
1 for EON45/EONSOFT
2 for EON42
3 for EON48
4 for EON48E
5 for EON48E Turret
6 for SETU VP248
Key Number is the number of the key which is to be reassigned the feature.
Key Numbers on EONSOFT are from 01 to 25
Key Numbers on EON42 are from 01 to 25
Key Numbers on EON48 are from 01 to 29.
Key Numbers on SETU VP248 are from 01 to 29.
For numbering of the keys, refer the default Key Maps illustrated at the beginning of this topic.
Function Number defines the exact function under each Function Type selected for the key.
Channel is the number of the channel in the BRI or PRI Line to be accessed.
Channel number for all function types other than BRI and PRI is 00
Function Function
Meaning Meaning
Type Number
00 None - -
01 SLT (Direct Station 001 to 512 Number of the SLT (software) port which is to be
Call) called when the key is pressed.
02 DKP (Direct Station 001 to 128 Number of the DKP (software) which is to be called
Call) when the key is pressed.
03 TWT (Direct Trunk 001 to 128 Number of the TWT (software) port which is to be
Access) called when the key is pressed.
04 BRI (Direct Trunk 001 to 032 Number of the BRI (software) port which is to be called
Access and Direct when the key is pressed.
BRI Channel
Access) Number of the Channel in the BRI Line, which is to be
accessed when the key is pressed.
05 T1E1 (Direct Trunk 001 to 008 Number of the T1E1 (software) port which is to be
Access and Direct called when the key is pressed.
PRI Channel
Access) Number of the Channel in the PRI Line which is to be
accessed when the key is pressed.
06 E&M (Direct Trunk/ 001 to 128 Number of the E&M (software) port which is to be
Station Access) called when the key is pressed.
12 QCKDIAL 001 to 999 The Global Directory Index number on which the
(Abbreviated External Number is stored. Call will be made to that
Dialing) External Number when this key is pressed
13 LOOP (Loop Count 01 to 10 The number of Call Appearance/Call Loop which will
or Call Capacity) be accessed when the key is pressed.
14 MACRO 01 to 25 The number string of the Key Board Macro which will
be dialed when the key is pressed. Refer “Key Board
Macro”
15 TAC (Direct Trunk 1 to 6 The Trunk Access Code that will be dialed when the
Access) key is pressed.
18 FEATURE (Direct 01 to 81 The feature that will be invoked when the key is
Feature Access) pressed. For a complete list of features, refer the
“Table: “Function Type 18: Features””.
26 SIP (Direct Trunk 01 to 32 The number of the SIP Trunk to be accessed when
Access) the key is pressed
29 Magneto (Direct 001 to 128 The number of the Magneto (software) port to be
Station Call) dialed when the key is pressed.
31 ROOM (Direct 001 to 512 The number of the Hotel Room to be dialed when the
Station Call) key is pressed.
33 Door Phone (Direct 1 to 3 The Door Phone port to be called when the key is
Station Call) pressed.
34 SIP Extension 001 to 500 The number of the SIP Extension to be dialed when
(Direct Access) the key is pressed.
Enter SE Mode 01
Enter SA Mode 02
Redial 07
Operator Dialing 12
Dynamic Lock 14
Hot Line 15
Alarm 16
Interrupt Request 18
Barge-In 19
Raid 20
Trunk Reservation 21
Call Toggle 22
Conference-Multi Party 25
Call Park 26
Room Monitor 28
Voice Help 30
Page Zone 33
Flashing on Trunk 38
Background Music 42
Meet Me Paging 43
Hot Desk 44
DND Override 45
Forced Release 49
Hold 50
Forced Answer 52
Maid Status 53
Guest Number 54
Minibar Details 55
Mute 56 Yes
Emergency Conference 57
SA Command 60
Floor Service 62
CLIR 64 Yes
Reminder 66
Opening a Door 75
RCOC Invoke 78
Check-In 001
Check-Out 002
Guest-In/Out 005
- 039
OG Print Filter: To Print calls of Duration more than this time 056
OG Print Filter: To Print calls of Units more than the units 057
programmed
Set filter to print internal calls with duration greater than that 070
given here
Program Day and Time for Internal Weekly Scheduled Reports 074
Program Date and Time for Internal Monthly Scheduled Reports 075
Set filter to print all calls with speech duration More than timer 083
Set filter to print all calls unanswered for duration More than 084
timer
Set filter to print all calls kept on hold for duration more than 085
timer
Set filter to print all IC calls recd. From nos. matching the 095
Number List
- 116
- 117
- 127
Flash 001 No
Pause 002 No
Release 007 No
Enter 009 No
Cancel 014 No
Answer 015 No
Examples:
Let us attempt to program the same sample Operator Template we customized earlier for EON48 using
a telephone. For this, we need to know the location of the key, i.e. the key number.
Existing function on the key To be replaced by Location of the key on the Key Map
To assign Trunk to Trunk Transfer to the key currently assigned to DKP4 key (key 01) in the Operator
Template, dial:
• 1261-1-1-3-01-18-24-00
Where,
1 is Operator Template
3 is for EON48 Terminal
01 is the Key Number
18 is Function Type for Features
24 is Function Number for Trunk to Trunk Transfer feature.
00 is Channel.
To assign Mobile Trunk 1 in place of SLT4 on DSS key 09 on the Operator Template, dial:
• 1261-1-1-3-09-25-01-00
Where,
1 is Operator Template
3 is for EON48 Terminal
09 is the Key Number
25 is Function Type for Mobile Trunks
01 is Function Number, i.e. the port offset of the Mobile Port.
To assign BRI Trunk 1 in place of SLT2 on DSS Key 11 on the Operator Template, dial:
• 1261-1-1-3-11-04-01-00
Where,
1 is Operator Template
3 is for EON48 Terminal
11 is the Key Number
04 is Function Type for BRI Trunks
01 is Function Number, i.e. number of the BRI port.
00 is Channel number.
To assign Channel 2 of T1E1 Trunk 1 in place of SLT1 on DSS Key 12 on Operator Template, dial:
• 1261-1-1-3-12-05-01-02
Where,
1 is Operator Template
3 is for EON48 Terminal
12 is the Key Number
05 is Function Type for T1E1 Trunks.
01 is Function Number, i.e. number of the T1E1 port.
02 is Channel number.
For example, to assign the customized Operator Template to DKP 001 and DKP002, you may dial:
• 1221-1-001-1 to assign the template to DKP001
• 1221-1-002-1 to assign the template to DKP002
OR
• 1221-2-001-002-1 to assign the same template to both DKPs at one go.
• Exit SE Mode.
When you personalize the Key Map of a DKP, you must first select the option 'Personalized' as the Key Map in the
DKP Parameters of the DKP.
You can personalize key maps of DKPs/Extended IP Phones using Jeeves or by dialing SE commands from a
telephone connected to the ETERNITY.
• Go to the DKP you want to assign a personalized key map, for example, DKP001.
• Select the option 'Personalized' as the 'Key Map' for the DKP.
• To assign features to keys, follow the same steps as you did for customizing the key templates.
• Click the key you want to assign the function. For example, you want 'Barge-In' on the key assigned to
DKP4, click this key.
• Click 'OK'. The window will close. The new label will appear on the key.
• When you have completed personalizing the key map, close the window.
• Follow the same steps to personalize the key map of another DKP.
For instructions on personalizing the Key Map of the Matrix Extended IP Phone, under Configuring SIP Extensions,
see “Matrix Extended IP Phone Settings”, “Phone Key Settings”.
For example, to assign the personalized Key map to DKP 003, dial: 1221-1-003-0
Key Number is the number of the key which is to be assigned the function/feature. The Key numbers
vary according to the EON Terminal Type being used.
Key Numbers on EONSOFT are from 01 to 25.
Key Numbers on EON42 are from 01 to 25.
Key Numbers on EON48 are from 01 to 29.
For numbering of the keys, refer the default Key Maps illustrated at the beginning of this topic.
Function Number defines the exact function under each Function Type selected for the key. Refer the
tables above for the complete list of function types and function numbers.
Channel is the number of the channel in the BRI or T1E1 PRI Line to be accessed.
Channel number for all function types other than BRI and PRI is 00
Channel number for all function types other than BRI and PRI is 00
For the complete list of Function Types and Function Numbers see “Customizing Key Templates using
a Telephone”.
The Key Numbers of the MATRIX Extended IP Phone is the same as EON48 described under this
topic.
Also see “Matrix Extended IP Phone Settings” under Configuring SIP Extensions using a Telephone.
• Exit SE Mode.
After you have finished assigning key templates and key maps, test the functioning of the keys.
If you want to use the proprietary DSS Consoles - DSS64 or DSS72 - with the EON, you must first install the DSS
Consoles and assign them to the respective DKP.
You can attach two DSS Consoles to a single DKP for further increase the number of keys. The two DSS Consoles
may be two DSS64 or DSS72 or a combination of DSS64 and DSS72. Refer “Direct Station Selection Console” to
know more.
For installation instructions, refer the topics Installing EON, Installing DSS Consoles, under “Installing ETERNITY
ME”, “Installing ETERNITY GE” and “Installing ETERNITY PE”.
For configuring DSS Consoles with DKPs, refer the topic “Configuring DKP Extensions”.
If you have attached the DSS Consoles - DSS64 or DSS72 - to the DKPs and assigned the DSS Consoles to the
DKP in the DKP Parameters, you may now program the DSS Console Keys.
01 17 33 49
01 25 49
02 18 34 50 02 26 50
03 19 35 51 03 27 51
04 20 36 52 04 28 52
05 21 37 53 05 29 53
06 22 38 54 06 30 54
07 23 39 55 07 31 55
08 24 40 56 08 32 56
09 25 41 57
09 33 57
10 26 42 58
10 34 58
11 27 43 59
11 35 59
12 28 44 60
12 36 60
13 29 45 61
13 37 61
14 30 46 62
14 38 62
15 31 47 63
15 39 63
16 32 48 64
16 40 64
17 41 65
18 42 66
19 43 67
20 44 68
21 45 69
22 46 70
23 47 71
24 48 72
In the default DSS Key assignment, all DSS Keys are assigned SLT ports for Direct Station Calling.
• If a single DSS64 is attached to a DKP, the SLT ports 005 to 068 are assigned to the keys. Recall that by
default SLT001 to SLT004 are assigned to the DSS keys on the DKP.
• If a single DSS72 is attached to a DKP, the SLT ports 005 to 076 are assigned to the keys. Recall that by
default SLT001 to SLT004 are assigned to the DSS keys on the DKP.
• If two DSS72 are attached to a single DKP, the SLT Ports 005 to 076 are assigned to DSS1 (the first DSS
Console) and the SLT ports 077 to 148 are assigned to DSS2 (the second DSS Console).
• If the first DSS is a DSS72 and the second DSS64, the SLT Ports 005 to 076 are assigned to DSS1 and
the SLT ports 077 to 140 are assigned to DSS2.
However,
• If two DSS64 are attached to a single DKP, the SLT ports 005 to 068 are assigned to DSS1 (the first DSS
Console) and SLT ports from 077 to 140 are assigned to DSS2 (the second DSS Console). The SLT ports
069 to 076 and from 141 to 148 cannot be accessed from the DSS. So, the keys of both DSS1 and DSS2
must be programmed.
• Similarly, if the first DSS is a DSS64 and the second DSS72, the SLT Ports 005 to 068 are assigned to
DSS1 and the SLT ports 077 to 148 are assigned to DSS2. The SLT ports 069 to 076 and from 141 to 148
cannot be accessed from the DSS. So, the keys of only DSS2 must be programmed.
The steps for programming the Keys of the proprietary DSS Consoles of Matrix, DSS64 and DSS72, are
quite similar the programming of the DKP keys. It can be done using Jeeves as well as a Telephone.
• The default key map of the DSS Console appears on your screen. By default all keys are assigned to
SLTs.
• The options for the Functions to be Performed by the key will open in a new window.
• Now, configure the key just like you configured the DKP/Extended IP Phone keys.
• Select the desired 'Function Type’ to be assigned to the key and the desired ‘Offset’ for the Function Type.
• Click ‘OK’.
• After you have finished assigning functions to the DSS1 keys, close the window.
• Follow the same steps as described above to program the keys of DSS2.
Key Number is the number of the key which is to be assigned the function/feature. The Key numbers
vary according to the type of DSS Console being used.
Key Numbers on DSS64 are from 01 to 64.
Key Numbers on DSS72 are from 01 to 72.
For numbering of the keys, refer the default DSS Key Maps illustrated at the beginning of this topic.
Function Number defines the exact function under each Function Type selected for the key.
Channel is the number of the channel in the BRI or PRI Line to be accessed.
Channel number for all function types other than BRI and PRI is 00
Refer the tables provided under “Customizing Key Templates using a Telephone” for the complete list
of function types and function numbers.
• Exit SE mode.
Depending on the number of BRI ports available to you and the type of Point-to-Multipoint Configuration (Short or
Extended Passive Bus), a maximum of 64 ISDN Terminals can be connected to the ETERNITY.
The number of ISDN Terminal extensions available to you for configuration also depends on the number of ISDN
Terminals you have specified on the“System Pre-requisites” page.
If you have enabled 'On-Site Configuration', the system will provide you only those ISDN terminal ports that are
actually present in the system for configuration.
Configure ISDN Terminal Parameters using Jeeves or by dialing commands from a Telephone.
• Configure the following parameters for each ISDN Terminal on this page:
• ISDN Terminal: This non-editable field is the number of the software port of the ISDN Terminal.
• Access Code: Assign Station Access codes to the ISDN Terminal. Station Access codes re commonly
referred to as Extension Numbers. These may be a combination of a maximum of 6 digits, which are
dialed to call the ISDN Terminal to which they are assigned.
All ISDN Terminal ports are assigned the following Station Access Codes as default.
01 3201
02 3202
03 3203
: :
: :
64 3264
• Name: Assign a 'Name' to the ISDN Terminal. The name may be of the person who will use the ISDN
Terminal or the name of a department. This name will be displayed to the called station.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the ISDN Terminal.
A Station Basic Feature Template includes a set of features that completely define the behaviour of the
Station, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call
Budget, and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all stations of the system that also
includes SLT ports, ISDN ports and E&M Lines with Station as Orientation Type.
Check if the default template fulfills the feature requirements (like “Class of Service (COS)”, “Toll
Control”, “OG Trunk Bundle Group”, etc.) of the ISDN Terminal.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all ISDN Terminals, retain Template 01.
If different sets of features are to be allowed to different ISDN Terminals, then prepare separate Station
Basic Feature Templates and apply them on the ports. To do this,
• Click the link 'Station Basic Feature Template' to open the page.
• Customize Template number 12 and click 'Submit' at the bottom of the page.
• Enter the number of the Template you customized, Template 12, in the 'Station Basic Feature
Template' field of the ISDN Terminal, for instance, ISDN-01, on which you want to apply this
• Repeat the same steps to customize and assign a different Template to another ISDN Terminal.
Also, refer the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on station ports.
• Station Advanced Feature Template: Assign a Station Advanced Feature Template to the ISDN
Terminal. The Advanced Feature Template consists of a set of less commonly used features like Message
Wait Notification Type, Alarm Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming
Call Routing, Storage of Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all stations of the ETERNITY, which
includes ISDN ports, SLT ports, and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the ISDN Terminal by clicking the 'Station
Advanced Feature Template' link.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to all
ISDN Terminals, retain Template 01.
If different sets of features are to be allowed to different ISDN Terminals, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
• Click the 'Station Advanced Feature Template' link to open the page.
• Enter the number of the Template you customized, Template 04 in the 'Station Advanced Feature
Template' field of the ISDN Terminal, for instance, ISDN-01, on which you want to apply this template.
If you want to apply this template to other terminals too, like ISDN-01, 02, and 03, assign the Template
04 to all these ports.
• Repeat the same steps to customize and assign a different Template to another ISDN Terminal.
Also refer the topic “Station Advanced Feature Template” for instructions on customizing these templates
and applying them on the station ports.
• Station Type: This parameter is relevant only for the Hotel Application of the ETERNITY. Extensions are
identified as 'Administrator' or 'Guest' extensions according to the intended user of the ISDN Terminal.
Advanced features
The above listed parameters fulfill the ISDN Terminal configuration requirements of most users. If you need to use
other features like Personal Directory, Call Pick-Up Groups, for some users, you may click the 'Advance' button.
• Personal Directory: Enter the number of the “Personal Directory” that you want to assign to the ISDN
Terminal. A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index
number (location code), Name and Trunk Access Codes (“OG Trunk Bundle Group”). The Personal
Directory is necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
The Personal Directory number that you assign to an ISDN Terminal must also be programmed either by
you, the System Engineer, or by the ISDN Terminal user. Refer the topic “Abbreviated Dialing” for
instructions on programming the Personal Directory.
• Priority: Select a Priority Level for the ISDN Terminal from 1, 2, 3... to 9, with '1' being lowest Priority and
'9' being highest Priority. Whenever a station (ISDN Terminal) with higher priority calls a station with lower
priority, a triple ring is placed on the called station. To know more, read the feature description “Priority”.
Call Pick Up allows the ISDN Terminal user to 'pick up' (answer) calls ringing on any other station, by
dialing a feature code, without physically going to the ringing station. For this to work, both the ringing
station and the station picking up the call must be in the same 'Call Pick Up Group'. Refer “Call Pick Up” for
instructions on how to create groups. You can create as many as 99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this ISDN Terminal in this field.
• If you have completed configuration of all the above listed ISDN Terminal Parameters, click 'Submit' at the
bottom of the page to save your changes.
• Exit SE mode.
In the context of a PBX, users understand the term 'Operator' as a person who handles multiple simultaneous calls
and functions as the link between callers and called parties.
For the PBX however, an 'Operator' is a Routing Group; a group of extensions to which calls made by extensions
by dialing '9' are to be landed. This also includes Direct Inward Dialing calls on trunks during which the caller dials
'9'.
Depending on the size of the Enterprise and the amount of call traffic to be managed, more than one Operator may
be employed. Also, it is not uncommon to have different Operator extensions according to the time of the day. For
instance, during working hours calls may be handled by the Receptionists or Front Desk Personnel, whereas during
non-working hours, calls may be handled by the Security Personnel.
To meet this requirement, ETERNITY offers configuration of up to four different Operators (Routing Group):
Operator 1, Operator 2, Operator 3 and Operator 4. However, at a time, only one Operator can be assigned to
extensions
Each 'Operator' is assigned a Routing Group for the Time Zones - Working Hours, Break Hours and Non-Working
Hours.
Each 'Operator' is assigned a Time Table, which defines the Working Hours, Break Hours and Non-working Hours
for a week. The system follows this Time Table to assign a Routing Group as 'Operator' according to the current
Time Zone.
1. Configuring Routing Groups as 'Operator': A routing group may be made up of one or more than one
extensions, depending on user requirement. If the user requires only one extension as 'Operator', include
only one extension as member in the Routing Group for Operator. If the user requires five extensions as
'Operator', create a Routing Group of the five desired extensions to be used as 'Operator'.
If the user requires Time-Zone based 'Operator', then prepare a different routing group for each Time
Zone. If the user requires the same Operator for all Time Zones, use the same Routing Group number in
all Time Zones.
2. Configuring a Time Table for Operator: This is applicable only when Operator extensions are different
for different Time Zones.
Define the Time Table to be followed for the Operator selection. The Time Table may be the same as the
Time Tables assigned to trunks and extensions of the ETERNITY or may be configured to match with the
timings of the persons who work as Operators. For example if Operators in the Enterprise are working in
shifts, the Time Table can be configured to match their timings.
3. Assignment of 'Operator' to Extensions: SLT, DKP, ISDN Terminals, and SIP phones connected as
extensions of the ETERNITY can be assigned to an 'Operator' in their Station Basic Feature Template.
All extensions may be assigned to the same Operator, or different groups of extensions may be assigned
to different Operators, so that call management is more efficient. For instance, certain extensions may be
assigned to Operator 1, certain others to Operator 2 and the rest to Operator 3.
4. Assignment of 'Operator' to Trunks: Trunks are also assigned an 'Operator', so that when a caller dials
'9' using Direct Inward Dialing (DID), the call is routed to the Routing Group defined as Operator for the
trunk for a particular Time Zone. For example, the during working hours, a caller on trunk 001 dials '9', the
call lands on 3001; when a caller on trunk 001 dials '9' during non-working hours the call lands on 3003
and when the caller dials '9' during break hours the call lands on 3002.
Decide the number of Operators to be configured on the basis of the user's requirement.
• Select a Routing Group you want to program for Operator. By default, Routing Group 32 is assigned to
Operator. You may configure this group, or select another one.
• Select the type of extension to be included in the group as 'Member Type'. The extension may be a DKP,
an SLT or an ISDN terminal.
If you have finished configuring Routing Groups for Operator, configure the Time Table for Operator.
• By default Time Table 1 is assigned to all Operators. If this time table meets your requirement, retain it. If
not, select another Time Table. Customize it by defining the Working Hours, Break Hours and Non-
Working Hours for the week.
• If you want to assign different Time Tables to different Operators, repeat the above steps to prepare the
other Time Tables.
• Select the Operator number you want to configure. By default Operator 1 is assigned to all extensions and
trunks.
• Select the number of the Time Table you prepared for the selected Operator.
• Enter the number of the Routing Group you prepared for the selected Operator for Working Hours, for
Break Hours and for Non-Working hours. If the same Routing Group is to be kept for all Time Zones, enter
the same number in fields of all three time zones.
Now, you may assign the 'Operator' groups you have configured to the SLT, DKP, ISDN Terminal, and SIP
extensions by configuring the number of the Operator (1-4) in the “Station Basic Feature Template” applied
on these extensions.
Similarly, you may assign the 'Operator' groups you have configured to the trunks in the “Trunk Feature
Template” applied on these Trunks.
For SE commands to assign an operator to extensions and trunks, refer the topics “Customizing
Station Basic Feature Template using a Telephone” and “Customizing Trunk Feature Template using a
Telephone”.
On issuing this command, Timetable 1 is assigned to Operator and Routing Group 32 is assigned to
Operator for all Time Zones.
• Exit SE mode.
• Two-Wire Trunks
• E&M Lines
• ISDN T1E1 PRI Lines
• ISDN BRI Lines
• Mobile Trunks
• SIP Trunks
• E&M Feature Template (for E&M Lines and T1E1PRI Lines that use E&M signaling)
Using these templates, you can configure all Trunks that are to be assigned the same set of hardware and software
features at one go, instead of configuring each trunk individually.
A TWT Hardware template must be assigned to all the TWT trunk ports. Using the TWT templates, you can
configure TWT ports which are to be assigned the same set of features at one go, instead of configuring port-by-
port.
The ETERNITY offers 50 TWT Hardware Templates. These templates have commonly used values, but can be
customized per the requirement and applied on the extensions.
TWT Type: Three types of TWTs can be interfaced to a TWT port of the ETERNITY:
• Normal Dial type: This is the conventional TWT available from the PSTN.
• Hotline type: The TWT connecting two destinations immediately on grabbing the trunk.
By default all the TWT ports are set as Normal Dial type. You may select the TWT Dial Type you want to
assign to the TWT port.
• AC Termination Impedance: The AC Termination Impedance of the TWT port must match with the AC
Termination Impedance supported by the PSTN network. The system supports the following AC
Termination Impedance:
• 600
• 900
• 270 + (750 || 150 nF) and 275 + (780 || 150 nF)
• 220 + (820 || 120 nF) and 220 + (820 || 115 nF)
• 370 + (620 || 310 nF)
• 320 + (1050 || 230 nF)
By default, the AC Termination Impedance is set as per the “Region” you have selected.
• Dial Type: You can select the Dialing method as Pulse or Tone (with configurable Pulse Ratio and DTMF
ON-OFF period) according to the Dialing method supported by the CO network to which the TWT port is
connected.
• Pulse Dial Ratio: This parameter is to be configured if you have selected Pulse as the 'Dial Type' in the
previous parameter. The system supports the six different Pulse Dialing Ratios on TWT ports. Select the
appropriate Pulse Dial Ratio from the following according to the type of Pulse Dialing Ratio supported by
your CO Network:
• 10PPS, 1:2
• 10PPS, 2:3
• 10PPS, 1:1
• 20PPS, 1:2
• 20PPS,2:3
• 20PPS, 1:1
By default, 10PPS, 1:2 is selected as the Pulse Dial Ratio.
• Rx CLI Type: ETERNITY detects the CLI sent by the CO network and sends this information to the
landing station/operator with the ringing signal. You must select the CLI Type supported by your CO
network from the following options:
• None
• Any ETSI DTMF format
• Any FSK V.23 format
• Any FSK Bellcore format
• 1st Ring, ETSI DTMF, 2nd Ring
• Polarity Reversal, ETSI DTMF, 1st Ring
• 1st Ring, FSK, 2nd Ring
By default, Any ETSI DTMF format is selected as the Rx CLI Type.
• Rx Gain (dB): You can increase or decrease the level of Incoming Speech (Receive Gain) on the Trunk by
changing the Rx Gain to the desired level from: 10, 9.5, 9, 8.5, 8, 7.5. By default Rx Gain is set to '0'.
• Tx Gain (dB): You can increase or decrease the level of Outgoing Speech (Transmit Gain) on the Trunk
by changing the Tx Gain to the desired level from: 10, 9.5, 9, 8.5, 8, 7.5 dB. By default Tx Gain is set to '0'.
This feature is particularly useful if you want to use “Call Cost Calculation (CCC)” to enable accurate
billing. When the signal is received, the billing will start and in the absence of this signal, the call will not be
billed, ensuring that unanswered and unsuccessful call attempts are not billed.
• Pseudo Answer: It is used when no signaling is available from the PSTN. If this option is selected, the
call will be considered as matured on the expiry of the 'Pseudo Answer Supervision Timer'
(configurable; default 20 seconds), irrespective of whether or not the call actually gets matured. After
this, the Call Duration Timer starts. Finally, the system starts detecting the “Disconnect Supervision”
signal configured for the TWT port.
• Polarity Reversal: It is used as maturity signal when the answer signaling is given in the form of
Battery Reversal. If the battery polarity of the line is -ve for TIP and +ve for RING, when the called party
has answered the call, the CO network will reverse the battery polarity, TIP becomes +ve and Ring -ve.
After this, the Call Duration Timer is started. Finally system starts detecting the Disconnect Supervision
signal configured for the TWT port.
• 12 KHz/16 KHz Pulse: When called party answers the call, the CO network generates a 12/16 KHz
metering pulse to indicate call maturity. Finally, the system will start detecting Disconnect Supervision
signal configured for the TWT port.
By default, Answer Supervision is set to 'Pseudo Answer' for each TWT port.
Select the same Answer Supervision signal as provided by your CO Network. If the type of Answer
Supervision signal selected in the system does not match with that of the CO network, the call will not be
stored in the Station Message Detail Record (SMDR) buffer. For example, if the CO network does not
support Answer Supervision, but you have set Polarity Reversal or 12/16KHz as Answer Supervision
Type, the call will be considered as matured and will not be stored in the Station Message Detail Record
(SMDR) buffer. The same would be the case if you selected 12/16KHz when the CO supports Polarity
Reversal.
• Pseudo Answer Supervision Timer: Configure this timer if you have selected 'Pseudo Answer' as
Answer Supervision Signal option.
This is the time period for which the system will wait before treating a call as matured (regardless of
whether or not it was answered). The range of this Timer is from 001 to 255 seconds. By default the
Pseudo Maturity Timer is set to 20 seconds.
When Pseudo Answer is selected as Answer Supervision signal, the call duration measured by the system
will not accurately reflect the actual call duration because the Pseudo Answer Supervision Timer is not
related to the actual call maturity. For example, if the Pseudo Answer Supervision Timer is set to 015
seconds, the call will be considered as matured after 015 seconds, even if it is answered after 20 seconds.
Similarly, if this Timer is set to 080 seconds, but the call was answered after 020 seconds and
disconnected after 040 seconds, this call will never be considered as matured as it ends before 080
seconds.
Disconnect Supervision signal is important when a PCO machine is connected to the (SLT Port)
ETERNITY and or when ETERNITY is deployed in a Gateway application.
In such application scenarios, it is desirable that calls that are disconnected by either end - calling party
or called party - is terminated by the system and the port is released. If the called (remote) party has
disconnected the call but the calling party (station that made the outgoing call from ETERNITY) has not
disconnected the call, the call remains live within the system.
So, Disconnect Supervision signal is important, particularly when calls are routed from TWT-to-TWT
ports, to indicate to the system that it needs to disconnect the call and release the port.
• None: When there this no signaling supported. Select this option only if there is no Disconnect
Supervision signal supported.
• Polarity Reversal: Call disconnection is signaled as Polarity Reversal when the call is
disconnected by the remote user. For example, if the battery polarity of the TWT port is '+ve' for TIP
and '-ve' for RING in speech condition then on disconnection by the remote user, TIP will become '-
ve' and Ring '+ve'. The user gets an Error tone and the TWT port is released.
• Open Loop Disconnect: Call Disconnection is signaled in the form of Open Loop, whereby the
Battery voltage on the TWT port is removed for a short duration. Voltage is restored after this short
duration. However, the Polarity of Battery Voltage on the TWT port is not changed.
This option is to be selected when call disconnection is signaled in the form of Open Loop
Disconnect pulse by the CO network. System will check Open Loop Disconnect signal for the time
configured for Open Loop Disconnect Timer for each TWT port. If the time of the Open Loop signal
detected is less than the Open Loop Disconnect Timer configured, it will not be considered as valid
Open Loop signal for releasing the TWT port. But if the Open loop is detected continuously for at
least for the time set as the Open Loop Disconnect timer, it is considered as a valid Disconnect
Supervision signal. The call will be released and caller will get error tone.
• Select the same Answer Supervision and Disconnect Supervision signal type supported by your CO
network for the TWT ports. Consider the following case:
• The CO network supports Polarity Reversal signal as Answer and Disconnect Supervision.
• But you have configured 'Pseudo Answer' as Answer Supervision signal and 'Polarity Reversal' as
Disconnect Supervision signal for the TWT ports in the system.
• In this case, when a call is made through the TWT port, the call will be considered as matured after
the Pseudo Answer Supervision Timer.
• Now, when the called party answers the call, the CO generates 'Polarity Reversal' as answer
supervision signal on the TWT port.
• Open Loop Disconnect Timer (ms): This parameter is applicable only if the option Open Loop
Disconnect is selected as Disconnect Supervision on the TWT port.
The range of this timer is from 001 to 999 milliseconds. By default, the Timer is set to 200 ms.
• DID Disconnection Tone Detection Timer: This parameter is relevant only for TWT Cards with CMX
Chipset. You need not configure this parameter if your TWT card hardware is DSP-based.
This parameter is to be configured when “Direct Inward Dialing (DID)” is enabled on the TWT, but the CO
network does not support Disconnect Supervision (Polarity Reversal or Open Loop Disconnect).
During a DID call, when DID is on, the system detects DTMF dialed by a caller. However, if the caller fails
to dial any digits, the system will wait for the duration of the First Digit Wait Timer. On the expiry of this
timer, the system will wait to detect the Disconnection Tone from the CO network. But since the CO
network does not support Disconnect Supervision or signaling (Polarity Reversal, Open Loop Disconnect,
Disconnect Tone), the system will not disconnect the DID call, even if the caller has disconnected it.
The DID Disconnection Tone Detection Timer prevents this by initiating Disconnection Tone detection for
the DID call. If the system detects the Disconnection Tone before the timer expires, the DID call will be
disconnected and the TWT port will be released.
If no tone is detected when the Timer expires, the DID call will be routed as per the Routing Logic
configured for the TWT port (Operator or Trunk Landing Group).
The range of this timer is from 1 to 9 seconds. By default the timer is set to 3 seconds.
• Disconnect Tone Detection: This parameter is to be configured if Call Disconnection is signaled by the
CO network in the form of Disconnect Tone.
When there is an incoming/outgoing call on/from the TWT port is answered, the system will check whether
the flag “Disconnect Tone detection - OG Call + All IC Call states other than DID No Digit Dial State” is
enabled.
Only if the flag is enabled, the system will detect the Disconnect Tone.
If Disconnect Tone is detected, the system will consider the call as ended and will release the TWT port.
• Disconnect Tone Cadence: To enable the system to detect the Disconnect Tone accurately, you must
set the Cadence (ON-OFF time) and Frequency of the Disconnect Tone, as supported by the CO network.
• Operator: This parameter has 3 options: No operator, Modulation (*), Addition (+). Default: No.
• Frequency 2 (Hz): Frequency 2 is from 20 to 1400 Hz. Select Frequency 2 if the Disconnect Tone
supported by the CO network is Dual Frequency. Default: 25Hz.
• ON Time 1 (ms), OFF Time 1 (ms): Select Cadence for the first cycle ON Time1 and OFF Time 1. It may
be 0, or 40 to 4000 milliseconds. Default: 750 ms ON Time 1, 750 ms OFF Time 1
• ON Time 2(ms), OFF Time 2 (ms): Select Cadence for the second cycle ON Time 2 and OFF Time 2. It
may be 0, or 40 to 4000 (ms). Default: 750 ms ON Time 2, 750 ms OFF Time 2.
• ON Time 3(ms), OFF Time 3 (ms): Select Cadence for the third cycle ON Time 3 and OFF Time 3. It may
be 0, or 40 to 4000 milliseconds. Default: 0 ms ON Time 3, 0 ms OFF Time 3.
• ON Time 4(ms), OFF Time 4 (ms): Select Cadence for the fourth cycle ON Time 4 and OFF Time 4. It
may be 0, or 40 to 4000 milliseconds. Default: 0 ms ON Time 4, 0 ms OFF Time 4.
When disconnect tone detected on the port matches the Frequency and Cadences you have set, the call
will be disconnected and the TWT port will be released.
When Disconnect cadence is zero, ETERNITY will skip that cadence and match the next cadence.
ETERNITY will match the cadence for 3 cycles and then release the trunk.
• Speech Delay Timer: It is the time after which the system gives dial tone to the extension, when the
extension user grabs the TWT.
To understand the significance of this timer, let us consider a situation. Extension 2001 does not have
calling permission for long distance numbers. The user of extension 2001 grabs a TWT, and dials a
number 1022-6305555. The system dials out this number, as it starts with '1', but since the actual dial tone
from the CO comes after some time, the CO interprets this number as 022-6305555 and establishes
speech. This way an extension user who does not have permission for long distance calling, can dial out a
long distance number. This situation can be prevented by setting the Speech Delay Timer to an
appropriate value.
The range of this timer is from 000 to 255 seconds. By default it is set to '1' second.
• Pause Timer: This Timer is used for the “Multi-Stage Dialing” feature.This Timer is required for inserting
delay while digits of a number string are out dialed from the TWT trunk. The range of this timer is from
0500 to 3000 milliseconds. By default the timer is set to 1000milliseconds.
To get accurate indication, the system supports Ring Cadence OFF timer on TWT port so that ring can
continue even for incoming calls with long Ring OFF period.
The range of the Ring Cadence OFF timer is from 1 to 6 seconds. By default the timer is set to 6 seconds.
• DTMF Out Dial: While dialing out the DTMF digits from the TWT port, the following attributes of DTMF
signal are critical.
• DTMF Signal ON Time (ms): It is the width of DTMF digit to be dialed out by the TWT port and is
configurable. By default the ON Time is set to 102 milliseconds.
• DTMF Inter-Digit pause Timer (ms): When the TWT port dials out the DTMF digits on the TWT, it
waits for the Inter Digit Pause Timer, while dialing the DTMF digits on TWT. This timer is configurable.
By default the timer is set to 102 milliseconds.
The 'level' of each DTMF digit is fixed, at -6.0 dB, but you may configure these parameters to match the
CO network requirement.
These DTMF Out Dial attributes are applied when the features Redial, Auto Redial and Abbreviated
Dialing are used to dial out the numbers from the TWT port. These attributes are also applicable when you
make a call from a DKP that has DTMF generation disabled.
• DTMF Detection: The default settings of DTMF Detection serve the requirements of most of the
applications. However, you may fine tune the following parameters if you face any problems in DTMF
detection.
• Minimum Level (dB): This parameter signifies the minimum level (dB) of the DTMF digit to be
considered as valid. By default, Minimum levels set to -4.5dB.
• Minimum ON Time (ms): This parameter signifies the minimum time period for which the DTMF signal
should be present in order to be detected. The valid range of this time is 17 to 204 milliseconds. By
default, Minimum ON Time is set to 34 milliseconds.
• Minimum OFF Time (ms): This parameter signifies the minimum time period between successive
DTMF digits. The valid range of this time is 17 to 204 milliseconds. By default, Minimum OFF Time is
set to 68.
• Flash Timer (ms): This parameter is relevant for dialing out Flash on the TWT to access some of the
features of the PSTN. Configure the desired time of Flash to be generated on the TWT.
• ON Hook Speed (ms): This parameter allows you to set the amount of time for the line-side device to go
on-hook.
The ON-Hook speed specified is measured from the time the ON-Hook bit is cleared until loop current
equals zero. Select the desired ON-Hook Speed from the following options:
• OFF Hook Speed (ms): This parameter defines the time to settle the line transients after which
transmission or reception can occur. Select the desired OFF-Hook Speed from the following options:
• 512 ms
• 128 ms
• 64 ms
• 8 ms
By default, OFF-Hook Speed is set to 8 milliseconds.
• Current Limiting: With this flag you can enable Loop Current Limiting mode. When this flag is enabled,
the Loop Current will be limited to a maximum of 60mA.
• Minimum Loop Current (mA): This parameter sets the minimum loop current at which DAA module of the
TWT port can operate. Select the minimum operational loop current from the following options as per your
requirement:
• 10
• 12
• 14
• 16
The minimum Operational Loop Current set by default is set to '10 mA'.
• Tip Ring Voltage (Volts): This parameter allows you TIP/Ring Voltage Adjustment on the line side.
Countries where Low voltage is required should use lower TIP/RING voltage. Adjust the values of the Tip
Ring Voltage to match your country requirements from the following options:
• 3.1
• 3.2
• 3.35
• 3.5
The default Tip/Ring voltage is 3.5.
• Ringer Impedance: Set the Ringer Impedance - High or Synthesized - for the TWT port according your
country-specific requirement.
'High' signifies 20Mohm Ringer Impedance. This is the default Ringer Impedance provided on the line side
by the DAA module of the TWT port. The DAA Module can provide higher impedance when 'Synthesized'
impedance is selected.
Some countries like Poland, South Africa and Slovenia require higher ring impedance which is achieved by
the DAA module, when Ringer Impedance is set to 'Synthesized' impedance.
• Ringer Threshold (Vrms): This parameter defines the level below which the TWT port would not validate
the Ring signal and the level above which it validate the Ring signal. Set Ringer Threshold to the desired
value from the following options:
• 13.5 - 16.5
• PPDC: 'Pre-PSTN Digit Count' or PPDC is parameter is to be configured, only if the TWT Trunk ports on
which the template is applied are in a “Behind the PBX Application”.
PPDC is the number of digits (dialed by an extension) to be ignored by the system before toll control check
is begun. It is the same as the number.
In Behind the PBX Applications, another PBX may be connected to the ETERNITY, with some of its TWT
Trunks terminating into the Stations of the other PBX and other trunks directly connected to the PSTN.
PPDC for TWT Trunk ports directly connected to the PSTN must be set to '0'.
For Trunk ports connected to stations of another PBX, PPDC must be configured as per the number of
digits in the Trunk Access Code defined for that PBX.
If the TAC is a single digit, select '1'. If TAC is double or triple digit number, accordingly select '2' or '3' as
the PPDC.
To know more about this feature, refer “Behind the PBX Application”.
• Gateway Application - Answer Signaling: This parameter is to be configured if the TWT Port is being
used in a gateway application as a source port (from where calls originate).
The calls originating on the source port (TWT port) are routed using another Trunk port, the terminating
port, which may be any trunk port, for example: T1E1. When call made from the terminating port gets
matured, this is signaled to the source port in the form of DTMF digits.
• Enable: Enable this flag if you want the TWT port to be used in a Gateway Application.
• DTMF String (max. 4 digits): Configure the DTMF digits to be sent to signal call maturity to the source
port.
• Category (Logical Partitioning): This parameter assigns the TWT Port to a trunk category for the
purpose of Logical Partitioning. By default all TWT Ports are assigned to Category 177.
If you have re-defined Category 1 or have assigned TWT ports to a different category, say Category 3,
enter the same number here.
You may configure the call permission between the Category assigned to TWT Ports and other Categories
(assigned to other Trunk ports). Refer the feature description “Logical Partition” to know more.
• Rx Gain at SIP Trunk (dB): This parameter allows you to increase the incoming speech volume level of
calls from SIP trunks to TWT trunks. By default, Rx Gain is set to 0dB.
• Tx Gain at SIP Trunk (dB): This parameter allows you to increase the outgoing speech volume level of
calls from SIP trunks to TWT trunks. By default, Tx Gain is set to 6dB.
77. Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network) are assigned this category.
• Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
• You can also set the Rx and Tx Gains for SIP to Digital Trunks and Stations and SIP to SLT.
• To increase Rx and Tx Gain for SIP to SLT stations, go to “SLT Hardware Template”. To increase Rx
and Tx Gain for Digital Trunks/Stations, go to SIP Trunk Parameters.
ETERNITY's versatile architecture allows it to be connected to such networks differing in their characteristics. You
can configure the TWT hardware features to match the standards supported by the POTS network of the country
where the system is installed.
The 50 TWT Hardware Templates offered by ETERNITY contain the default values of the above-listed parameters.
The default parameter values of these are country specific and are loaded in each template according to the
Country selected as the“Region”.
For example, when India is selected as the Region, the default value of the AC Impedance is 600, whereas it is
900 ohms when Philippines is selected as Region, and 320 + (1050 || 230 nF) for Region UK. Similarly, the
default Rx CLI Type for Region India is 'Any ETSI DTMF format' while the same is 'Any FSK V.23 format' for Region
UK.
By default, TWT Hardware Template number 01 is assigned to all TWT Trunk ports.
While the default TWT Hardware Template number 01 has all with commonly used values to match your country-
specific requirements, you can still customize each of the 50 Templates to match your preference or requirement.
If the TWT Hardware Template number 01 fulfills your requirements, and if the same features at to be applied on all
TWT trunk ports, retain Template 01. Similarly, if you want only a few changes to be made to Template 01 and
apply it on all TWT Ports, make the changes and retain the template.
However, if different sets of features are to be allowed to different TWT hardware ports, then prepare separate TWT
Hardware Templates and apply them on the ports as required.
You can use Jeeves or a Telephone to customize the TWT Hardware Template.
78. Normally, fax calls require less gain compared to voice calls. However, to improve fax reception, ETERNITY allows the configuring
of gain settings for fax. Fax gain settings consist of Data Gain and Bypass Gain. ETERNITY supports Fax Receive Gain for SIP to
TWT Trunks, SIP to Digital Trunk calls as well as SIP to SLT Calls.
• Click the link 'TWT Hardware Template' under TWT Configuration to open the page.
• Select a TWT Hardware Template Number you wish to customize. For example, Template 07.
• Go to the TWT software ports to which this Template you customized (07) is to be assigned, for example
TWT-007 and 008.
• Repeat the same steps to customize another template and apply it to the TWT Port.
Table 2:
For example, to change the Rx CLI Type in Template 07 from 'Any ETSI DTMF format' to 'Any FSK V.23
format', dial 5902-1-07-01-3
Where,
07 is the template number
05 is the parameter number for Rx CLI Type
2 is the code for Any FSK V.23 format.
• Exit SE mode.
A Trunk Feature Template is assigned to all the Trunk types: TWT, Mobile, SIP, BRI, T1E1PRI, and E&M.
A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours, Non-Working
hours for a week.
Certain features of the ETERNITY like Operator, DID, DISA, Trunk Landing Group, require the trunk to
behave differently in each Time Zone. For example, it can be made to land on the Operator Station during
working hours, and on the station of the dining area during Break (lunch) hours, and on the station of the
Security Personnel during non-working hours.
So, a Time Table is assigned to stations defining the Time Zones for the entire week, so that the system
can execute the Time Zone-dependent features and facilities according to the Time Table.
There are 8 different Time Table templates to select from. By default, the Time Table 1 is assigned to all
Trunk Feature Templates. All seven days of the week are 'working hours 9:00-18:00' with break hours
'13:00 -14:00 hrs'.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the
Trunk Feature Template. Please refer the topic “Time Tables” for more details.
• Operator: Define the Operator for the Trunks on which the template is applied. Operator is used to route
the call when the caller dials '9' during a DID call. This parameter is of significance only if DID/DISA is
enabled on the trunk.
The system supports multiple Operators. In each Time Zone any one of the four Operators can be
selected.
Trunks may be assigned to a single Operator, or different groups of Trunks may be assigned to different
Operators, so that call management is more efficient. For instance certain Trunks may be assigned to
Operator 1, while some may be assigned to Operator 2 and the rest to Operator 3.
Operator 1 is the default in the Trunk Feature Template. If you want to assign different trunks to different
Operators, you must create a separate Trunk Feature Template with a different Operator for each trunk
group.
To configure the TLG, you must first configure Routing Groups. Refer “Trunk Landing Group (TLG)” for
instructions on configuring trunk landing groups. Also refer the topic “Routing Group”.
There are as many as 96 Routing Groups which can be assigned as TLG. By default, Routing Group 01 is
assigned as TLG for all Time Zones. If you have prepared a different TLG for each Time Zone, for
example, Routing Group 02 for Working Hours and Break Hours, Routing Group 3 for Non-Working Hours,
then enter the number of these Routing Groups in the TLG field.
• DID: This parameter is to be configured if you want to enable “Direct Inward Dialing (DID)” on the trunk
ports on which you will apply the template.
DID can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break Hours (BH) and
Non-Working Hours (NH).
For each Time Zone, you may select the desired DID option from the following:
• OFF: Select this option if you want to disable DID for the Time Zone. By default, DID is disabled (OFF)
for all the Time Zones.
• ON: Select this option if you want the calls to be answered by the built-in Auto Attendant of the
ETERNITY. ETERNITY answers the call using Voice Modules, if assigned, or it answers the call and
plays the appropriate call progress tone - Dial tone, Ring Back tone, Busy tone - for each call state.
If you select this option, make sure you also configure the DID related Timers and Flags, record and
assign the DID related Voice Message. Refer the topics “Direct Inward Dialing (DID)” and “Voice
Message Applications” for instructions.
• Delayed DID: Select this option if you want to the calls to be answered by the built-in Auto Attendant of
ETERNITY, if none of the extensions of the Trunk Landing Group answers the incoming call.
DID Delayed Timer: Set this Timer, if you want to enable “Delayed DID” on the trunk. The range of this
timer is from 01 to 99 seconds. By default, the timer is set to 10 seconds.
When you enable Delayed DID, ETERNITY routes the incoming call on the trunk to the Trunk Landing
Group assigned to this trunk. It waits for the duration of the Delayed DID Timer for any of the extensions
the Trunk Landing Group to answer the call.
If none of the extensions in the Trunk Landing Group answers the call before the expiry of the DID Delayed
Timer, the call is answered by the built-in Auto Attendant of ETERNITY.
• Multilevel DID Profile: Select the desired Multilevel DID Profile from 1 to 16, if you have enabled
Multistage DID for a time zone. By default, 1 is selected.
• DISA: This parameter is to be configured if you want to enable “Direct Inward System Access (DISA)” on
the trunk ports on which you will apply the template.
DISA can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break Hours (BH) and
Non-Working Hours (NH).
• PIN Auth.-Multiple calls: Select this option if you want to enable DISA with PIN Authentication and
allow multiple calls during the DISA login session.
• CLI Auth.-Multiple calls: Select this option if you want to enable DISA login with CLI Authentication
and allow multiple calls to be made during the DISA login session.
Caller numbers that do not match with the CLI Table will be routed as per the logic of the Trunk Feature
Template.
• CLI Auth.-One call-Answer Signaling: Select this option if you want to enable DISA session with CLI
Authentication, and allow only a single call to be made during the DISA login session. This form of
DISA is used when ETERNITY is installed in a Gateway application. This form of DISA is applicable on
TWT Trunks only.
• Trunk Auto Answer: This parameter is relevant only if you want to enable the “Trunk Auto Answer”
feature on the Trunk ports on which this template is applied.
Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller
with a voice message before the call is actually handled.
• Disable: Select this option if you do not want Trunk Auto Answer on the trunk.
• For all Calls: Select this option if you want all incoming calls landing on the trunk line to be answered.
• When Busy: Select this option if you want the system to answer incoming calls on the trunk to be
answered if the landing destination is busy.
Trunk Auto Answer can be enabled or disabled for each Time Zone, namely Working Hours (WH),
Break Hours (BH) and Non-Working Hours (NH).
If you have enabled Trunk Auto Answer for All Calls or When Busy, you must also set the Trunk Auto
Answer Greeting Message.
• Trunk Auto Answer Greeting Message: This parameter is to be configured only if you have enabled
Trunk Auto Answer 'For All Calls' or 'When Busy' for a Time Zone in the previous parameter.
Assign the number of the Trunk Auto Answer Greeting with which callers will be greeted. For this you must
first record a Voice Module with the desired Greeting Message and assign it to this parameter.
You can assign up to 4 Greetings Messages for Trunk Auto Answer and assign a different Greeting
message for each Time Zone. Refer the topic “Voice Message Applications” for instructions on configuring
the greetings.
• Trunk Auto Answer RBT Message Type: This parameter is to be configured only if you have enabled
Trunk Auto Answer 'For All Calls' or 'When Busy' for a Time Zone in the previous parameter.
When Trunk Auto Answer is enabled on a trunk, the system will answer the caller with a Greeting message
once, and play the Ring Back Ton (RBT) Message Type you have selected. You can select an RBT
Message Type from the following options:
• None: The system will play Ring Back Tone to the caller after the Trunk Auto Answer Greeting
Message.
• Internal MOH: The system will play internal music-on-hold to the caller after playing the Trunk Auto
Answer Greeting Message.
• External Music: The system will play music to the caller from an external source connected to the
Analog Input Port.
If you select this option, you must also connect an External Music source to the Analog Input Port. You
may refer to the Installation Instructions for your model of ETERNITY.
• RBT Message: The system will play a voice message continuously to the caller.
If you select this option, you must first record a Voice Module with the desired RBT Greeting Message.
You can set up to 4 RBT Messages. You can also assign a different RBT message for each Time Zone.
Refer the topic “Voice Message Applications” for instructions on configuring the RBT Message.
Assign the number of the RBT Message you want to be played to callers in each Time Zone.
By default, 'None' is selected as RBT Message Type for all Time Zones.
• Trunk Auto Answer Busy Bye Message: This parameter is to be configured only if you have enabled
Trunk Auto Answer ('For All Calls' or 'When Busy') for a Time Zone.
When Trunk Auto Answer is enabled on a trunk, the system will answer the caller with a Greeting message
once, and play the Ring Back Tone (RBT) Message Type you have selected (see previous parameter)
continuously for the duration of the DID Inactivity Timer. If the landing destination (called extension) is busy
on the expiry of this Timer, the system will inform the caller about the busy state in two ways, which you
can select from the following options:
• Bye Message: The system will play a voice message to the caller.
If you select this option, you must first record a Voice Module with the desired Bye Message.
You can set up to 4 Busy Bye Message. You can also assign a different Busy Bye message for each
Time Zone. Refer the topic “Voice Message Applications” for instructions on configuring the Busy Bye
Message.
Assign the number of the Bye Message you want to be played to callers in each Time Zone.
• Priority: Select a Priority Level for the trunks on which the template will be applied.
Each trunk of the ETERNITY can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority.
Whenever there are incoming calls on multiple trunks, the call on the trunk with higher priority will be
answered by the Operator station first. To know more, read the feature description “Priority”.
By default, the Priority of all trunks is set to '9'. Decide what Priority Level you will assign to the trunks and
set the desired level for the trunk.
• SMDR-OG Storage: This flag is used to enable or disable the storage of details of outgoing calls from the
trunk. Please refer the topic “Station Message Detail Recording-Storage” for more details. By default,
storage of outgoing calls is enabled.
• SMDR-IC Storage: This flag is used to enable or disable storage of details of incoming calls on the trunk.
Please refer the topic “Station Message Detail Recording-Storage” to know more. By default, storage of
incoming calls is enabled.
• Hold on DSS Key Press: This flag defines the 'Hold' state of the external called party, when an extension
user presses a DSS key to dial another port.
For example, the DKP extension user (on DKP-001 port) is in the middle of speech with an external party
on a Trunk port TWT-002.
If extension user of DKP-001 presses a DSS key to call another extension port DKP-003, two situations
are possible, depending on whether the Hold on DSS Key Press flag is enabled or disabled:
• When the Hold Flag is enabled: TWT-002 will be played music-on-hold. DKP-001 will hear Ring Back
Tone and the call will be placed on DKP-003.
• When Hold Flag is disabled: TWT-002 will be disconnected. DKP-001 will hear Ring Back Tone, and
call will be placed on DKP-003.
• Forced Account Code Flag: This parameter is related to the “Account Codes” feature of the ETERNITY.
This flag must be enabled, if the feature Forced Account Code is to be applied on the trunks.
When this flag is enabled, the system will prompt extension users to dial the Account Code whenever they
grab a trunk to dial out a number. The system will allow extension users to dial out numbers only when
after they have dialed the Account Code or Name.
By default, the flag is disabled. Refer the feature description for “Account Codes” to know more.
Account Codes feature must also be enabled in the Class of Service of extension users who are to be
allowed this feature.
To use this feature, you must first configure a Routing Group as FAX TLG. Refer the topic “Routing Group”
for instructions.
You may configure any Routing Group number as Fax TLG. By default, Routing Group 01 is assigned as
Fax TLG. As the same Routing Group is used for other feature applications such as Trunk Landing Group,
Floor Service, Alarms, you are recommended to configure a different Routing Group number as Fax TLG.
Now, assign the number of the Routing Group you have prepared as Fax TLG.
• Call Cost Calculation Pulse Rate Option: This parameter is to be configured only if you want to apply
the “Call Cost Calculation (CCC)” feature on the trunks on which the template is applied.
You have four options for Pulse Rate Types. Select from Pulse Rate Type for Pulse Rate Option 1 to 4
which you want to apply on the trunks.
• Call Cost Calculation Time Schedule: This parameter is to be configured only if you want to apply the
“Call Cost Calculation (CCC)” feature on the trunks on which the template is applied.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
must first define the Time Zone (time of the day) for which a particular Pulse Rate should be applied and
the Time Schedule for each Time Zone.
You can configure up to four different Time Zones - T1, T2, T3 and T4 with different Pulse rates in the
CCC-“Holiday Pulse Rate Table”.
Now, configure the Call Cost Calculation Time Schedule, by specifying the Start Time and the End time (in
24hours: minutes format) for each Time Zone.
The default Time Schedule (starts and end time) for each Time Zone Index are as follows:
T1 00:00 23:59
T2 00:00 23:59
T3 00:00 23:59
T4 00:00 23:59
If your service provider offers the same Pulse Rate for the entire day,
• configure only one Time Zone Index with the Pulse Rate, for instance, T1, in the CC-Holiday Pulse
Rate Table.
• Now, set the Time Schedule for Time Zone, T1, with the start and end time in Hours: Minutes format;
• set the start and end time of the other Time Zone Index, T2 to T4, to 00:00 (hours: minutes).
Similarly, if your service provider supports two different Pulse Rates in a day, set the Start and the End time
for two Time Zones and set the other two to 00:00.
If the default values of the Template fulfill the requirements of all Trunk types, retain this template. If you want to
change some of the feature settings and apply the template to all trunk types, you may simply customize this
template.
However, if you want to assign different feature settings for different trunk types, you are recommended to prepare
and apply separate Trunk Feature Templates for each Trunk type.
• Apply the Trunk Feature Template you customized to the Trunk Port type.
• Go to the TWT software ports to which this Template you customized is to be assigned, for instance
TWT-001 to 003.
• Enter the number of the Trunk Feature Template you customized, 02, in the 'Trunk Feature Template'
field of this port.
• Repeat the same steps to create another template and apply it on the desired Trunk Port type.
To apply the template on E&M ports, open 'E&M Parameters' page under E&M Configuration.
To apply the template on SIP Trunks, open 'SIP Parameters' page under VoIP Configuration.
To apply the template to BRI Trunks, open 'BRI Parameters' page under BRI Configuration.
To apply the template on T1 lines, open 'T1 Port Parameters' page under T1E1 Configuration.
To apply the template on E1 lines, open 'E1 Port Parameters' page under T1E1 Configuration.
• 5802-*-Feature Number-Code to set the same value for the parameter in all templates.
Template Number is the number of the Trunk Feature Template from 01 to 50.
Parameter Number is the number of the Trunk Feature Template Parameter from 01 to 35.
Code is the value for each parameter from 0 to 9.
Refer the following table for the parameter numbers and the values for the codes.
• Exit SE mode.
A SIP Hardware Template must be assigned to all SIP Trunks as well as SIP Extensions. Using the SIP Hardware
Template, you can configure SIP Trunks and Extensions the same set of features at one go.
ETERNITY offers 32 SIP Hardware Templates, which can be customized as per the requirement and applied on
SIP Trunks and Extensions.
• Vocoder Preference: Vocoders are the various voice codecs used to compress the data in RTP packets
for optimum use of bandwidth and for ensuring voice quality. You can set 7 Vocoder options in the order of
preference for a SIP trunk.
The Vocoders supported by the VoIP card of the ETERNITY in the order of preference, from 1st to 7th, by
default are: G.723, G.729ab, GSM FR, iLBC - 30 ms, iLBC - 20 ms, G. 711-Law, and G. 711 A-Law.
• G.723 Bit Rate (kbps): You can select the Bit Rate for G.723 codec as 5.3 kbps or 6.3 kbps. When G.723
is negotiated, the selected Bit Rate will be applied only when sending the RTP packets. When receiving
RTP packets from the remote end, both Bit Rates of G.723 will be accepted. The default G.723 Bit Rate is
6.3kpbs.
• Silence Suppression for Vocoders: This parameter suppresses the 'Silence' packets, allowing only the
Voice packets through. ETERNITY supports Silence Suppression for all Vocoders except GSM FR.
Silence Suppression must be disabled if you have selected 'Pass Through' as the “Fax Type”.
• Rx Gain for SIP to Digital Trunk/DKP Voice call (dB): This parameter allows you to increase the
incoming speech volume level of calls from SIP trunks to Digital trunks (BRI, T1E1, and Mobile).
• Tx Gain for SIP to Digital Trunk/DKP Voice call (dB): This parameter allows you to increase the
outgoing speech volume level of calls from SIP trunks to Digital trunks (BRI, T1E1, and Mobile)
• You can also set the Rx and Tx Gains for SIP to Analog Trunks and Stations (TWT and SLT).
• To increase Rx and Tx Gain for SIP to TWT trunks, go to “TWT Hardware Template”.
• To increase Rx and Tx Gain for SIP to SLT stations, go to “SLT hardware Template”.
• DTMF Type: This parameter will determine how the DTMF digits will be sent over the IP Network, when a
DTMF digit is pressed. The ETERNITY supports three DTMF options: i) RTP (RFC 2833), ii) SIP Info, and
iii) In-Band. You may select the appropriate option. By default RTP (RFC 2833) is selected.
• Echo Cancellation: ETERNITY supports Echo Cancellation for SIP to TWT trunk calls and SIP to Digital
Trunks (BRI, T1E1, Mobile, SIP) and Stations (DKP, ISDN Terminals). If you want to apply Echo
Cancellation, you must enable configure the following parameters.
• Enable: This flag is to be enabled to apply Echo Cancellation on SIP to TWT and SIP to Digital Trunks/
Stations. By default Echo Cancellation is enabled.
• Tail Length (msec) for TWT: Echo Cancellation Tail Length for SIP to TWT trunks can be 32/64/128
milliseconds. By default, Echo Cancellation Tail Length for TWT is set to 64 milliseconds.
• Tail Length (msec) for Stations and Digital Trunks: Echo Cancellation Tail Length for SIP to Digital
Trunks/Stations can be 32/64/128 milliseconds. By default, Echo Cancellation Tail Length for Digital
Trunks/Stations is set to 32 milliseconds.
• Jitter Buffer: The speed at which voice packets travel through a network depends on the condition of the
network. All voice packets may not come at the same speed. This variation in the delay in receiving
packets, known as Jitter, affects voice quality. Jitter Buffer helps overcome the delay in receiving voice
packets and improves voice quality. Jitter Buffer receives voice packets, stores them and sends it to the
DSP to process it at evenly spaced intervals, thus improving voice quality.
• Type: Select the type of Jitter Buffer - Static or Dynamic - you want to use. If you selected Static Jitter
Buffer, you may set the 'Minimum Delay'. The value configured in the Minimum Delay determines the
size of the Static Jitter buffer.
If you selected Dynamic Jitter Buffer, configure the 'Optimization Factor' and 'Minimum Delay'. The
minimum size of the Dynamic Jitter buffer depends on the 'Minimum Delay' you configured. The
Optimization Factor determines the rate of adaptation of the Dynamic Jitter Buffer to the jitter in the
network.
• Optimization Factor: This parameter must be configured if you have selected Dynamic Jitter Buffer.
The Optimization Factor can be set from 01 to 13. By default, it is set to '10'.
In networks with higher jitter, a higher value should be set as Optimization Factor.
The actual size of the Dynamic Jitter Buffer will be determined by the DSP on the basis of the value of
the Optimization Factor and actual network condition. Dynamic Jitter buffer can go up to maximum 300
milliseconds.
• Minimum Delay (msec): This parameter is to be configured for both Static and Dynamic Jitter Buffer.
The Minimum Delay determines the size of the Static Jitter Buffer and When Jitter Buffer type is
selected as Static, the Minimum Delay defines the size of the Static Jitter Buffer. The Static Jitter Buffer
will store each received voice packets for the time you set and then it will send it to DSP for voice
processing.
When Jitter Buffer type is Dynamic, the Minimum Delay specifies the minimum time for which the
Dynamic Jitter Buffer will store the received voice packet before sending it to the DSP for voice
processing. 'Minimum Delay' can be from 10 to 280 milliseconds. By default Minimum Delay is set to
150 milliseconds.
• Fax Type: This parameter allows you to select the protocol of Fax over IP (FoIP). You can send/receive
Fax from a Fax machine connected to the SLT port of the ETERNITY.
The ETERNITY VoIP Card supports the fax options: T.38 (UDPTL), T.38 (RTP), and Pass Through.
• 'Pass Through' and 'T.38' will work only if the peer devices also support the same option.
• If you select 'Pass through' as Fax type, you must disable 'Silence Suppression'.
• If the fax sent using T.38 is rejected, ETERNITY will use Pass Through for sending the Fax.
• T.38 Fax Parameters: This parameter is relevant only if you have selected T.38 as the Fax Type of Fax
over IP. Receiving a good quality fax over SIP trunk depends on high 'Eye Quality Monitor' (EQM). The
higher the Eye Quality Monitor, the better the Fax quality. To improve the quality of T.38 fax reception, you
may configure the below parameters.
• Max Rate (Kbps): This parameter controls the Fax image transfer speed. As EQM is inversely
proportional to Fax Max Rate, if you receive poor quality fax, the Fax Max Rate should be reduced. The
default Max rate is 14.4 kbps.
When you reduce the Fax Packet Period, the fax image will be sent at lower speed. EQM is inversely
proportional to Fax Packet Period.
The default packet period is 40msec. Do not change the default settings unless it is required.
• Image Redundancy Level: The Fax Image Level is redundancy level for output Image, which can be
from 0 to 3.
Fax Image transfer speed is inversely proportional to this parameter. If this parameter is low then fax is
transferred faster. EQM is directly proportional to this parameter. If this parameter is high, good quality
fax can be achieved.
You may increase the Fax Image Level from 1 to 3 if the quality of fax does not improve with Fax Max
Rate and Fax Packet Period.
• Data Redundancy Level: This is a redundancy level for T.30 control data. Fax Data Level can be set from
0 to 7. Level 0 means no redundancy. Redundancy level increases from 1 towards 7. The higher the level
set, the slower would be the fax transmission.
EQM is directly proportional to this parameter. The higher the Fax Data Redundancy Level, the better the
EQM. By default, Data Redundancy Level is set to 3.
• Pass Through Fax: This parameter is of relevance if you have selected 'Pass Through' as the Fax type.
Pass Through Fax packets are transported using RTP protocol. Normally, fax calls require less gain
compared to voice calls. However, to improve fax reception, ETERNITY allows the configuring of gain
settings for fax. Fax gain settings consist of Data Gain and Bypass Gain.
ETERNITY supports Fax Receive Gain for SIP to Digital Trunk calls as well as SIP to SLT Calls.
• Pass Through FAX Rx Gain (SIP-Digital Trunk Call): Configure this parameter if Pass Through Fax is to
be received on a Digital Trunk (Mobile, BRI, T1E1PRI).
• Data Gain (dB): select the dB Level for Data Gain. By default, Data Gain is set to -11 dB.
• Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
• Pass Through FAX Rx Gain (SIP-SLT Call): configure this parameter if Pass Through Fax is to be
received on a fax machine connected to an SLT port.
• Data Gain (dB): select the dB Level for Data Gain. By default, Data Gain is set to -11 dB.
• Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
If you want to change the values of certain SIP Hardware Parameters, but apply the same parameter values to all
SIP Trunks and Extensions, simply customize the desired parameters in Template 01.
If different hardware parameters are to be applied to different SIP Trunks and SIP Extensions, customize different
the SIP Hardware Templates using Jeeves or a Telephone and apply them to the SIP Trunks and SIP Extensions
as appropriate.
• Select a Template number you wish to customize, for example Template 02.
• Now, apply this Template 02 on the SIP Trunks and SIP Extensions.
• Go to the SIP Trunks to which this Template is to be assigned, for example SIP Trunk 02, 03 and 04.
• Enter the number of the Template you customized, 02, in the field 'SIP Hardware Template' of each of
these SIP Trunks.
• Go to the SIP Extension software ports to which the Template is to be assigned, for example SIP
Extensions-005 to 008.
• Enter the number of the Template you customized, 02, in the field 'SIP Hardware Template' of each of
these SIP Extensions.
• Repeat the same steps to customize another template and apply it on the SIP Trunks and Extensions.
For example, to select T.38 (RTP) as Fax Type in Template 02l, dial 7806-1-02-19-2
Where,
02 is the template number
19 is the parameter number for Fax Type
For example, to apply SIP Hardware Template 02 on SIP trunks 02, 03, and 04, dial 7808-2-02-04-02
For example, to apply SIP Hardware Template 02 on SIP Extensions 005 to 008, dial 7807-2-005-008-
02
• Exit SE mode.
The E&M Feature Template is applied also on T1E1PRI trunks which use E&M signaling.
The ETERNITY offers 50 such Templates. The E&M templates are loaded with default values that serve the
requirements of a broad user base, but can be customized as per user requirements.
• Seizure Type: E&M Trunk Seizure Type is the line protocol that defines how the equipment seizes the
E&M Trunk. It is also referred to as Start Dial Supervision Signaling Protocol. ETERNITY supports the
following Seizure Types:
Outgoing Call:
While making an outgoing call, when the station user of PBX A seizes the E&M Port of PBX A, the
status of the "M" wire of its E&M port will go high, indicating that it has seized the E&M line.
There will not be any signaling over the "E" wire of PBX A's E&M Port during seizure.
Incoming Call:
While receiving an incoming call over its E&M port, PBX-B will be ready to receive digits as soon as it
detects high state on its "E" wire.
There will not be any signaling over the "M" wire of E&M Port of PBX B while receiving an incoming
call.
• Immediate with Ack: The method of seizing the E&M Line using Immediate with Acknowledgement for
Outgoing and Incoming calls is as follows:
Outgoing Call:
If this seizure type is selected, while an outgoing call is made by seizing the E&M Port, the 'M" wire will
go high immediately.
The remote end will acknowledge this by making its 'M' wire high, which in turn will activates (high) 'E'
wire of the E&M port of PBX-A.
On sensing high signal on 'E' wire, PBX-A will start dialing out the DTMF/Pulse digits.
Incoming Call:
On detecting high signal on "E" wire of the E&M port, the system will consider it to be an incoming call
seizure and hence it will immediately make its 'M" wire high, which will allow the remote end to dial out
the DTMF/Pulse digits.
Call Disconnection:
If the parameter 'Release Type' is configured as 'Status Change' and for this type of seizure, the "M"
wire at the remote end goes Low for some call condition, the call will be disconnected. For example,
"M" wire at the remote end will go 'Low' in the following conditions:
• Remote end user dials invalid number and does not hang up on getting Error Tone.
• Remote end user dials valid station number and after conversation remote end hangs up first.
• Remote end user dials valid number and station does not reply, but the remote party does not
disconnect the call.
• Immediate + Wink: The method of seizing the E&M Line using Immediate + Wink Start for Outgoing
and Incoming calls is described below.
0V 0V
Wink
-48V -48V Pulse
2001 3001
M M
E E
Rx Rx
PBX-A PBX-B
Tx Tx
SA SA
2002 3002
SB SB
Outgoing Call:
While making an outgoing call when the PBX A attempts a seizure (grab), the state of the "M" wire of
the E&M port of PBX A will go high.
To acknowledge this, the E&M port of PBX B will send Wink signal over its "M" wire, when PBX B is
ready to receive digits.
PBX A will wait for the duration of the 'Wait Wink Timer'. On receiving the acknowledgment in the form
of Wink signal on the "E" wire of its E&M port, before the expiry of the Wait Wink Timer, PBX A will
consider the trunk seizure as successful. Digits will be dialed out from E&M port.
If Wink is not received from PBX B within the ‘Wait Wink Timer’, PBX-A will drop the call.
Incoming Call:
While receiving an incoming call over the 'M' wire of its E&M port, PBX B will send the Wink signal to
the PBX A, which has initiated the seizure (grab).
The ‘Wink’ signal will be sent by the PBX B when it is ready to receive the digits from the PBX A.
You can change the 'wink' pulse width by configuring the 'Wink Pulse Timer'.
The width of the Wink Pulse ranges from 0000 to 9999 milliseconds.
Outgoing Call:
When PBX-A attempts a seizure of (attempts to grab) the E&M Port, the 'M' Wire of the E&M Port of
PBX-A will go high and wait for the E&M Port of PBX B to turn its 'M' wire high.
PBX-B detects this on its 'E' wire. To acknowledge this, the E&M port of PBX-B will turn its 'M' wire high
and send a Wink signal over its 'M' Wire. Sending of the wink signal indicates the readiness of PBX-B
to receive the digits of the called party number.
PBX-A will wait for the duration of the 'Wait Wink Timer' to receive the acknowledgment in form of the
Wink Signal on the 'E' wire of its E&M port.
When PBX-A receives the acknowledgment from PBX-B, before the Wait Wink Timer expires, PBX-A
considers the trunk seizure as successful and starts dialing out DTMF digits as per the MFCR2
Signaling protocol.
However, if PBX-A does not receive the Wink Signal within the ‘Wait Wink Timer', or if invalid Wink
Pulse is received (not according to Wink Pulse Timer), PBX-A will drop the call by turning its 'M' wire
low.
Incoming Call:
On detecting high signal on 'E' wire of the E&M port, PBX-B will consider it to be an incoming call
seizure and hence it will immediately make its 'M' wire high and send the Wink signal on the 'M' wire of
PBX-A to indicate its readiness to receive the called party number digits. The width of the Wink Pulse
(referred to as 'Proceed To Send' in MFCR2 Signaling) can be changed by setting the 'Wink Pulse
Timer'.
PBX-A dials out the digits as per the MFCR2 Signaling protocol.
When the called station of PBX-B answers the call, PBX-B sends the Wink signal on the 'M' wire of its
E&M Port to indicate the call maturity.
Call Disconnection:
The call can be disconnected by the calling party, PBX-A, or the called party, PBX-B by changing the
status of the 'M' wire to Low.
Call Disconnection takes place when 'M' wire is low. So, it is recommended that the Call 'Release Type'
of the E&M Port for this Seizure Type (Immediate with Ack+Wink) be set to 'Status Change'.
• Seizure Pulse: The method of seizing the E&M Line using Seizure Pulse for Outgoing and Incoming
Calls is described below.
Outgoing Call:
While making an out going call from the E&M port of PBX-A, it will send Seizure Pulse over the "M" wire
of its E&M port to seize the line.
Incoming Call:
While receiving an incoming call over the E&M Port PBX B will detect valid Seizure Pulse over the "E"
wire of its E&M port.
Seizure Pulse can be set for various time periods T1, T2 and T3 as required.
The Seizure Pulse for T1, T2 and T3 ranges from 000 to 999 milliseconds.
• Seizure Pulse + Wink: The method of seizing the E&M Line using Seizure Pulse + Wink for Outgoing
and Incoming Calls is as follows:
Outgoing Call:
While making an OG call, "Seizure Pulse" (as configured) will be sent on the "M" wire of E&M Port and
will start ‘Wait Wink Timer’ and expect ‘Wink’ from the remote device.
On receiving a valid Wink Pulse from the remote end within the Wait Wink Timer, digits will be dialed
out on the E&M Port.
If a valid Wink Pulse is not received from the remote end, digits will be dialed out on the expiry of ‘Wait
Wink Timer’.
Incoming Call:
On detecting valid Seizure pulse (matching with configured value of seizure pulse) on "E" wire of the
E&M Port, the E&M Port will send Wink pulse (of configured value) on "M" wire, and the call will be
considered to be present.
To make a call from PBX A to PBX B, the caller from PBX A presses the DSS Key of the desired E&M
port.
For as long as the DSS key is pressed by caller from PBXA, the signal on the "M" wire of E&M port of
PBX A will be high.
When the destination station on PBX B answers, the caller from PBX A releases the DSS key, as the
line seizure is successful.
When the caller from PBX A releases the DSS key, the signal on the 'M' wire of the E&M port of PBX A,
and the "E" wire of the E&M port of PBX B will go low.
• Radio A: This seizure type to be used for supporting Combat Net Radio (CNR) signaling on the E&M
port.
The E&M port of the ETERNITY will detect this pulse on the 'E' wire of its E&M port and recognize it as
a seizure signal (incoming call indication).
The length of this input signal (pulse) can be defined by setting the 'Minimum Pulse Width for Radio
Seizure' Timer. The ETERNITY will recognize the input signal on the 'E' wire of its E&M port as a
seizure signal only if the signal is present for the duration of this timer.
Once the incoming call is detected by the E&M port, the call is routed to the Routing Group number as
per the call routing logic configured for the E&M port.
When Routing Group member (Station) answers the call, speech is established with the CNR user.
Outgoing Call:
When any station user of the PBX wants to contact the CNR user (Soldier), station user must grab the
E&M port by dialing TAC/Selective Trunk Access / DSS key of E&M Port.
When E&M Port is grabbed by the station user, the 'M' wire of the E&M port is made high, to indicate
the seizure signal to the radio equipment. The radio equipment then passes on the call to the CNR
user's wireless phone.
Incoming Call:
When status of "E" wire of the E&M port goes high for greater than or equal to the 'Minimum Pulse
Width for Radio Call' it is considered to be an incoming call, and the call is routed as per the current call
routing logic.
When a station user of the PBX answers an incoming call, the 'M' wire of the E&M Port will be made
high.
The call between the station user and the CNR user can be disconnected only if the station user
disconnects the call. So, it is recommended that the call 'Release Type' of the E&M port be set to
'None'.
• Radio B: Characteristics of M and E wire for seizure type 'Radio B' are as follows:
Outgoing Call:
When any station user of the PBX wants to contact the CNR user (Soldier), station user must grab the
E&M port by dialing TAC/Selective Trunk Access / DSS key of E&M Port.
When E&M Port is grabbed by the station user, the 'M' wire of the E&M port is made high, to indicate
the seizure signal to the radio equipment. The radio equipment then passes on the call to the CNR
user's wireless phone.
Incoming Call:
When the status of "E" wire of the E&M port goes high for greater than or equal to 'Minimum Pulse
Width for Radio Call' ETERNITY considers it to be an incoming call, and routes the call as per the
current call routing logic.
When a station user answers the call, there is no signaling on the 'M' wire of the E&M port by
ETERNITY.
The call between the station user and the CNR user can be disconnected only if the station user
disconnects the call. So, it is recommended that the call 'Release Type' of the E&M port be set to
'None'.
The following table shows the status of the "M" wire of an E&M Port while making Outgoing calls and
while receiving Incoming calls.
Select 'Station' if the E&M port is to function as Station. All Station-related parameters, the “Station
Basic Feature Template” and the “Station Advanced Feature Template”, will be applicable to this port.
Select 'Trunk' as orientation type if the port is to be assigned the feature of a trunk line. All trunk-related
parameters, “Trunk Feature Template”, will be applicable to this port.
Select 'Tie Line'79 if the E&M port is to function as both a Station and a Trunk. The system will regard
the port as a Station for incoming calls and as a Trunk for outgoing calls. The Station Basic and
Advanced Feature Templates as well as the Trunk Feature Template will be applied on this port.
• Dial Type: Digits can be dialed over E&M Tie Line by two methods:
• Tone: In this Dial Type, the DTMF signals will be sent on the "Tx" of the E&M port of the originating
side and it will be received over the "Rx" of the E&M port of the terminating side.
• Pulse: In this Dial Type, the dialed digits will be sent on the "M" wire of the E&M port of the
originating side and will be received over the "E" wire of the E&M port of the terminating side.
The way digits are sent varies according to the Trunk Seizure Type, as described for each Seizure Type
below.
• Express: the caller can make call by pressing the DSS key.
• Immediate: the PBX seizes the Tie Line and the system will start the Pause Timer. On the expiry of the
Pause Timer the PBX sends the digits to the remote PBX.
• Seizure Pulse + Wink: on receiving Wink signal from the terminating end, the originating side (which
initiates seizure) will start the Pause Timer and on expiry of Pause Timer it will start sending digits.
• Seizure Pulse: the originating PBX system will start the Pause Timer after sending the Seizure Pulse.
On expiry of the Seizure Pulse Timer it will send digits to the terminating end.
• Pulse Dial Ratio: This parameter is to be configured if 'Pulse' is selected as Dial Type in the previous
parameter. Select the Pulse Dial Ratio from any of the following values:
• 10PPS, 1:2
• 10PPS, 2:3
• 10PPS, 1:1
• 20PPS, 1:2
• 20PPS, 2:3
• 20PPS, 1:1
The default Pulse Dial Ratio is 10PPS, 1:2
• Wait Wink Timer (sec): This parameter is to be configured, if 'Immediate + Wink' or 'Seizure Pulse +
Wink' have been selected as Seizure Type for the E&M port.
The Wait Wink Timer is the Time period for which the system waits for the acknowledgement in the
form of Wink Signal on "E" wire of the E&M port of the ETERNITY to consider it as a successful
seizure.
The range of this timer is from 000 to 255 seconds. By default the timer is set to '000' seconds.
The Wink Pulse Timer defines the width of the Wink Pulse. The range of this timer is from 0000-9999
milliseconds. By default the timer is set to '0000' milliseconds.
• Seizure Pulse Timers (msec): This parameter is to be configured, if 'Seizure Pulse' is selected as
Seizure Type for the E&M port.
• T1: This is the time period of the first ON period of the 'Seizure Pulse'.
• T2: this is the time period of the second ON period of the 'Seizure Pulse'
• T3: this is the time period of the third ON period of the 'Seizure Pulse'.
The range of T1, T2 and T3 is from 000-999 milliseconds. By default the timer is set to '000'
milliseconds.
• Minimum Pulse Width for Radio Seizure (msec): This Timer is to be configured if 'Radio A' or 'Radio
B' have been selected as the Seizure Type for the E&M ports.
The Minimum Pulse Width for Radio Seizure defines the time for which the ETERNITY will wait to
detect the width of the pulse sent by the Radio Interface Device80 on the 'E' wire of the E&M port and
recognize it as a seizure signal (incoming call indication).
The range of this timer is from 000 to 999 milliseconds. By default the timer is set to '150' milliseconds.
• Release Type: ETERNITY supports four methods to 'release' E&M calls based on, which end will
release the call and 'release pulse width'. These are:
• None: Select this option if you have selected "Express" as Trunk Seizure Type for the E&M port. It
is advisable to keep the Release Type as "None" in case the protocol does not support any
signaling for disconnecting the E&M port, for the reason that 'Trunk-to-Trunk Inactivity Timer' will be
started if the E&M port with Release type "None" is involved in a Trunk-to-Trunk call.
• Release Pulse: Select this option if the specific Pulse width of the Release Pulse is to be used to
disconnect the call. This Pulse width is configurable, as shown below:
The call can be disconnected by either party by sending Release Pulse. Consider the following
example:
80. Connected on the 'M' wire of the E&M port. The Radio Interface Device sends pulse of approximately 100msec or higher.
• Status Change: Select this option, if Status change of 'M' wire (‘M’ wire is low) is to be considered for
release of the E&M call.
By default, 'Status Change' is selected as the Release Type for all E&M ports.
If you selected ‘Immediate with Ack + WInk’ as Seizure Type for the E&M Port, select ‘Status Change’
as Release Type for the port.
• Release Pulse Timer (msec): This timer is to be configured if you selected the option 'Release Pulse'
as the Release Type for E&M calls in the previous parameter.
This timer defines the specific Pulse width of the Release Pulse is to be used to disconnect the call.
The range of this timer is from 0000-9999 milliseconds. By default the timer is set to '0000'
milliseconds.
• CCS - When End Point: CCS (Compander Control Signal) is a type of signal used by PLCC Networks
to improve the quality of speech transmission. The PLCC network awaits this signal from the PBX
when speech is established. ETERNITY supports CSS. The system sends CSS signal to the PLCC
panel.
This parameter is relevant if the E&M line is being used in a Power Line Communication Network
(PLCC).
This flag should be enabled if the E&M port is used as an Endpoint in a PLCC network. When the E&M
Port is used as an Endpoint, the system sends CSS to the PLCSS panel while making an outgoing call
through the E&M port and when receiving an incoming call on the E&M port.
• CCS - When Transit Exchange: This flag is to be enabled if the E&M port is used as a Transit
exchange in a PLCC network.
When the E&M Port is used as a Transit Exchange: The system sends CSS to the PLCC panel when
there is an Incoming/Outgoing Transit call through the E&M port.
• Max. OG Pulse Digit Count: This count defines the maximum number of digits that can be dialed out
to make a call. When dialing out the number, if the number of digits exceeds this count, the port which
is used for dialing these numbers is released automatically.
• Idle Wait Timer (sec): This timer signifies the time after which the codes could be simply (Station
Numbers or Station Numbers with Exchange ID) dialed over the E&M trunk.
• When Forced Disconnection is used. For example, two exchanges A and B are connected
through E&M trunk. Station 2002 of PBX A is talking to station 3001 of PBX B over the E&M line.
Station 2001 of PBX A calls station 3001 of PBX B and finds it to be busy.
Station 2001 is allowed to use forced disconnection feature. Station 2001 issues the forced
disconnection command. The PBX A disconnects the Station 2002. It then waits for the Idle Wait
Timer to expire and then dials 3001 over the E&M trunk.
• To stop any station from grabbing the E&M trunk until call is released. For example, when
Station 2001 of PBX A goes On-Hook, PBX A sends a release signal over the E&M trunk to PBX B.
In turn, PBX B sends a release signal to PBX A as an acknowledgment.
The E&M Idle Wait Timer set in PBX A does not allow any other station of PBX A to grab the E&M
trunk. Similarly, E&M Idle Wait Timer set in PBX B does not allow any station of PBX to grab the
E&M trunk.
• Flash Timer (msec): This Timer is significant when the PBX acts as a Transit exchange for a call. The
flash received on one E&M Port is generated on another E&M Port involved in a Transit call.
• Pause Timer (msec): This Timer defines the time for which the system waits before dialing the outside
number after grabbing the E&M trunk.
Sometimes a station user may not get a dial tone immediately on grabbing a trunk, in which case the
station user may wait for the dial tone before dialing out the number. However, when the system dials
out the number, if there is no pause time, it is possible that the system may dial out the number before
getting the dial tone. This may result in a wrong number being dialed out. The Pause Timer helps avoid
this.
• Pseudo Answer Supervision Timer (sec): This is the time period after which, the system will
consider the call as matured, irrespective of whether the call was answered or not. At the end of the
Timer, the system will start detecting Disconnect Supervision.
The range of this timer is from 000 to 255 seconds. By default the Timer is set to 030 seconds.
• Ring Timer (sec): This is the time for which the stations connected to ETERNITY ring for incoming
calls.
The Ring Timer is useful in situations where the users may not be able to immediately answer on the
first few rings. The range of this timer is from 000 to 255 seconds. By default the Timer is set to 255
seconds.
The range of this timer is from 000 to 999 milliseconds. By default the Timer is set to 750 milliseconds.
• DTMF Out Dial: This parameter is of significance when the system dials out DTMF digits to enable the
device at the remote end (in this case a PBX) to detect and decode the Tones. You must configure
both the DTMF ON Time and the Level (dB) according to the DTMF digit detection capacity of the
remote PBX.
For example, PBX A and PBX B are connected over E&M Line. PBX B detects DTMF digits only if the
tone remains present (ON) for 100 milliseconds frequency and at a transmit level of 4 dB. The DTMF
Out Dial parameter for PBX A should be configured accordingly. The DTMF Out Dial ON Time should
be set to 100ms and Level to 4dB.
• ON Time (msec): This is the Time for which the DTMF digit tone will remain ON, while being dialed
out by the ETERNITY. The range of this timer is 50 to 500 milliseconds. By default the ON Time is
set to 100 milliseconds.
This Timer must be configured according to the DTMF digit detection capacity of the remote device.
• Level (dB): This is the Transmit Level of the DTMF digit dialed out by the system. The range of
DTMF Out Dial 'Level' is from 0 to 7. By default DTMF Out Dial Level is set to 3.
• MFC R2 Signaling: This parameter is relevant only if you selected 'Immediate with Ack + Wink' as the
Seizure Type. Configure the following timers related to MFCR2 Signaling.
• Forward Tone Maximum ON Time (T1) (sec): the range of this timer is from 1 to 99 seconds. By
default it is set to 15 seconds.
• Forward Tone Maximum OFF Timer (T2) (sec): The range of this timer is from 1 to 99. By default
it is set to 24 seconds.
• Maximum Compelled Cycle Timer (T3) (sec): the range of this timer is from 1 to 99. By default it
is set to 15 seconds.
• Pulse Duration for Pulse Signal (msec): The range of this timer is from 001 to 999. By default, it
is set to 150 seconds.
• Pulse Signal Maximum Wait Timer (sec): The range of this timer is from 1 to 99. By default, it is
set to 15 seconds.
• First Forward Tone Wait Timer (sec): The range of this timer is from 8 to 24. By default, it is set to
15 seconds.
• Minimum MF Signal Persist Timer (msec): The range of this timer is from 1 to 255 seconds. By
default, it is set to 20 seconds.
If you have re-defined Category 3 or have assigned E&M ports to a different category, say Category 2,
enter the same number here.
You may configure the call permission between the Category assigned to E&M Ports and other
Categories. Refer the feature description “Logical Partition” to know more.
If all the E&M Ports are to be configured as 'Stations', then retain this template.
If all the E&M Ports are to be configured as 'Trunks', use the default E&M Feature Templates 09 and 10 which have
'Trunks' as Orientation Type.
If some of the E&M Ports are to be configured as Stations, some as Trunks and yet others as Tie Lines, prepare
different E&M Feature Template for each Orientation Type and apply them to the related ports.
The E&M Feature Template can be customized using Jeeves and a Telephone.
81. Trunk ports used to interconnect two PBXs are assigned this category.
• Now, apply the E&M Feature Template you customized to the E&M Ports and T1E1PRI ports.
• Enter the number of the Template you customized in the field 'E&M Feature Template' of the selected
T1E1 trunk port.
Refer the following table for the Parameter Number and default parameter values of all E&M Feature
Templates.
For example, you want to Immediate with Ack +Wink in the E&M Template number 2.
• Dial 6002-1-2-01-9
Where
2 is for E&M Template number 2
01 is parameter number for Seizure Type
9 is the value of Immediate with Ack+Wink
• Exit SE mode.
The ETERNITY supports a maximum of 128 Analog Two-Wire Trunk Lines82. Before you begin configuring the
TWT trunk ports, ensure that the TWT card has been installed correctly.
You may configure the TWT ports from Jeeves and using a Telephone.
• TWT Port No.: This non-editable field is the number of the software port of the TWT Trunk.
• Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the TWT Card has been
inserted. 'Port' is the number of the TWT trunk port on that card.
82. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
• Enable Port: This flag is for enabling or disabling a TWT Trunk port. When a TWT Trunk port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clearing this check box.
You may disable TWT port in case of trouble with the TWT line.
• Name: You may assign a 'Name' to each TWT Trunk to facilitate identification. Whenever there is an
incoming call without CLI on this port, the Name you have programmed will be displayed on the landing
extension.
The Name of the port may be the name of the Service Provider of this Trunk Line (recommended).
• TWT Hardware Template: A TWT Hardware Template is a set of features that completely define the
behavior of the hardware port of the TWT, such as Type of TWT Trunk, AC Termination Impedance, Pulse-
Tone Dialing, Answer Supervision, Disconnect Supervision, DTMF detection, etc.
Apply a TWT Hardware template to the TWT trunk port. The ETERNITY offers 50 TWT Hardware
Templates. By default, TWT Hardware Template number 01 is assigned to all TWT Trunks. Refer the topic
“TWT Hardware Template” to know more.
Check if this default template fulfills the feature requirements of the TWT Hardware Ports by clicking the
'TWT Hardware Template' link.
If TWT Hardware Template 01 fulfills your requirements, and if the same features at to be applied on all
TWT trunk ports, retain Template 01. Similarly, if you want only a few changes to be made to Template 01
and apply it on all TWT Ports, make the changes and retain the template.
However, if different sets of features are to be allowed to different TWT hardware ports, then prepare
separate TWT Hardware Templates and apply them on the ports as required. To do this,
• Apply the TWT Hardware Template you customized to the TWT Port by entering the template number
in the 'TWT Hardware Template' field of this port.
To know more about the hardware port features and customizing templates, refer the topic “TWT
Hardware Template”.
• Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator, DID,
DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the behavior of a
Trunk. Apply a Trunk Feature Template to the TWT Trunk port. By default, Trunk Feature Template 01 is
applied on all TWT Trunks as well as all other trunk types like ISDN BRI, ISDN T1/E1/PRI, GSM, and VoIP.
Refer the topic “Trunk Feature Template” to know more.
Click the 'Trunk Feature Template’ link to open the page. Check if the default Template 01 fulfills your
requirement for the TWT Trunk port.
If the default Template 01 does not fulfill your requirement, you may prepare a different Trunk Feature
Template and apply on all TWT Ports. For this,
• Go to the TWT Software Port Number you want to assign the Template you prepared.
• Enter the number of the Template you prepared (02) in the 'Trunk Feature Template' field.
You may also prepare different Templates for different TWT Ports, for example Template 02 for certain
ports, Template 03 for others. In which case, follow the steps described above. For each TWT Port, enter
the number of the template you have prepared for that port.
To know more about customizing templates, refer the topic “Trunk Feature Template”.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the TWT
Trunk port.
Cost Factor is used to grade the cost of routing calls from a TWT trunk, from 1 to 99; where 1 denotes least
cost and 99 denotes highest cost.
Assign a Cost Factor to the TWT Trunk port, for instance 02, and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the TWT Trunk Port
001 only,
• You must first assign a Cost Factor (01-99) to TWT Port 002, for example, 02.
• Enter '6' in the 'Number' column, Cost Factor '02' as Preference 1, 2, 3 and 4.
All outgoing calls assigned Cost Factor trunk 02 will be made from TWT Trunk Port 002.
• Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this TWT trunk port:
• Type: Select the type of Call Budget on Trunk—Amount, Minutes or Number of Calls—to be applied on
this TWT trunk port. By default, no Call Budget type is selected.
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
• Calls: If you selected ‘Calls’ as the Call Budget Type, enter the number of calls in this field. By default,
the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes to be reset on a
particular date of every month.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
• Call Back: This parameter is related to the ‘Call Back on Trunk Port’ feature. If you want to enable the 'Call
Back on Trunk Port' feature on this TWT trunk, configure the following parameters:
• Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
• Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote party
should be routed:
• DID
• PIN Authentication - Multiple Calls
• CLI Authentication - Multiple Calls
• CLI Authentication - Single Call - Answer Signaling
• Operator
• Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the ‘CLI number”. If you want the call back to be made to a different number, select ‘Alternate
Number’.
• Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all TWT trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all TWT trunks and Trunk Port types, retain
this list.
If you want a different set of number strings to be programmed for this TWT Trunk, select a different
Number List, and assign it to the TWT trunk port.
You may program the Incoming Number List either from the ‘Number List’ page or by clicking the
‘Incoming Number List’ link to reach the Number List page.
Refer the topic “Number List” to know more, and for configuration instructions.
• Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the ‘Incoming Number List’, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all TWT trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this TWT Trunk port.
You may program the Outgoing Number List either from the ‘Number List’ page or by clicking the
‘Outgoing Number List’ link to reach the Number List page.
Refer the topic “Number List” to know more, and for configuration instructions.
• Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an Outgoing Trunk Bundle Group (OTGTBG).
Select ‘Same port’ if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
• OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTBG that you define here for Call Back.
• If you have completed configuration of all the above listed TWT Parameters, click 'Submit' at the bottom of
the page to save your changes.
• For commands to program Call Budget on TWT trunks, refer the topic “Call Budget on Trunk”.
To select Call Back From port for a TWT Trunk port, dial:
• 3316-1-TWT-Call Back From to select Call Back From for a single TWT trunk.
• 3316-2-TWT-TWT-Call Back From to select the same Call Back From for a range of TWT trunks.
• 3316-*-Call Back From to select the same Call Back From for all TWT trunks.
Where,
TWT is the number of the software port of the TWT trunk from 001 to 128.
Call Back From is
1 for Same Port
2 for OGTB Group
Default: Same Port
• Exit SE mode.
The ETERNITY supports a maximum of 64 Mobile ports83. Before you begin configuring the Mobile ports, ensure
that the GSM/3G card has been installed correctly.
2. Network Selection
You can customize mobile port parameters and select the network using Jeeves and a Telephone. However,
mobile port status can be viewed using Jeeves only.
83. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
• Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the mobile card has been
inserted. 'Port' is the number of the Mobile port in which you have installed the SIM card.
The number of the Universal Slot will depend on the model of ETERNITY you are using. The number of
the Mobile Port depends on the configuration of the Mobile card. For instance, if you have installed
GSM8 card, the number of the ports will be from 1 to 8.
By default the ETERNITY can detect and assign the hardware slot and port numbers automatically to
the mobile (software) ports. However, if required, you may change the Hardware Slot and Port
assigned to the mobile software port. In which case, enter the desired Hardware Slot and Port number
in this field.
If you want to de-assign the Hardware Slot and Port, enter '00' in both fields.
• Enable Port: This flag is for enabling or disabling a Mobile Trunk port. When a Mobile Trunk port is
disabled, neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clicking the check box.
• Name: You may assign a 'Name' to each Mobile Trunk to facilitate identification. Whenever there is an
incoming call without CLI on this port, the Name you have programmed will be displayed on the landing
extension.
The Name of the port may be the name of the Service Provider of the SIM Card you have installed on
this port (recommended) or the phone number assigned to the SIM card on this port.
• Band Selection (MHz): The Frequency Band supported by the mobile networks varies from country to
country. ETERNITY's mobile card supports frequency bands of most countries. Select the Frequency
Band used by your GSM/3G Provider for the mobile port.
Frequency Band selection not required if the Mobile Card has SIMCOM 3G module.
When you change the Frequency Band, the change will be effected after the next system restart or the
next Mobile Port restart.
• SIM PIN: If you have enabled PIN protection and changed the SIM PIN of the card to the default value
'1234' using a mobile handset84, you can assign a new PIN to the SIM card from the ETERNITY.
Make sure that the PIN stored on the SIM card and that of the system are the same.
The SIM PIN will not be set to default value, when you restore the default settings of the system.
If you have enabled PIN protection, and the SIM PIN on the Card and the SIM PIN programmed in the
ETERNITY are not the same, the SIM card may get blocked and would require the Personal Unblocking
Number (PUK) from the Service Provider to reactivate it again.
84. Refer the topic Installing the Mobile Card for instructions. If you have not enabled PIN protection before installing the GSM card,
you will not be able to change the SIM PIN.
• Accept': incoming calls will be allowed and incoming call logic is applied.
• ‘Ignore': incoming calls will not be processed further and call logic will not be applied.
• ‘Reject': incoming calls will be rejected immediately and mobile port will be freed (released).
By default, incoming calls are accepted. This feature is particularly useful in outbound call centers for
blocking incoming calls on a SIM number (mobile port) of the ETERNITY.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the
mobile port.
Cost Factor is used to grade the cost of routing calls from a mobile trunk, from 1 to 99; where 1 denotes
least cost and 99 denotes highest cost.
Assign a Cost Factor to the mobile Trunk port, for example, 03 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '9' through the SIM installed in
Mobile Port Number 01 only,
• You must first assign a Cost Factor (01-99) to Mobile Port 01, for example, 03.
• Enter '9' in the 'Number' column, Cost Factor '03' as Preference 1, 2, 3 and 4.
All outgoing calls assigned Cost Factor trunk 03 will be made from Mobile Port 01.
• Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator,
DID, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the behavior of
a Trunk. Apply a Trunk Feature Template to the Mobile Trunk. By default, Trunk Feature Template 01
is applied on all Mobile Trunks as well as all other trunk types. Refer the topic “Trunk Feature
Template” to know more.
Click the 'Trunk Feature Template’ link to open the page. Check if the default Template 01 fulfills your
requirement for the mobile port.
If the default Template 01 does not fulfill your requirement, prepare another Trunk Feature Template85,
and enter the newly prepared Template number for the Mobile port.
• CLIR: Enable this flag if you want to activate CLIR for all outgoing calls made through the Mobile Port.
When CLIR is enabled, the called party will not be able to see the subscriber number of the Mobile
Port.
85. The default template is applied on the ports of all trunk types supported by ETERNITY. Changes to the default template will be
applied on all trunk types also. So, you are advised to prepare a new template and apply it to the desired trunk types.
This feature will work only if subscribed/supported by your mobile service provider.
• Return Call to Original Caller (RCOC): Enable this flag if you want to apply the RCOC feature.
If this feature is enabled on the Mobile trunk port, the system routes calls returned by remote parties
back to the extensions that originally made the call from this port (the original callers' extensions). To
know more, refer the feature description for Return Call to Original Caller (RCOC).
Advanced Configuration
The above listed parameters fulfill the basic mobile trunk port configuration requirements of most users. However, it
is anticipated that some users may need to configure other less commonly used features on the mobile ports, such
as Call Budget, Call Back on Mobile Port, or they may want to use the Mobile port as a Gateway application.
For such users, you may click the 'Advance' button and program the following parameters:
• N/w Registration Retry Count: The mobile port is programmed to automatically locate and register with
the Network that supports the SIM card installed on it. Also, at each power ON, the mobile port (SIM) will
automatically register with the Network that supports the SIM on it.
However, if the Mobile port fails to register, it will restart the process of network registration on the expiry of
the Network Registration Retry Timer86. On the expiry of this timer, the system will retry registration for the
programmed Count (number of times) and with each re-try attempt, the count will be decremented by one.
• Tx Gain: You can increase or decrease the Transmit volume level from the mobile port by increasing or
decreasing the Transmit Gain (Tx) of the Mobile port. By default Tx is set at 3.
• Rx Gain: You can increase or decrease the Receive volume level on the mobile port by increasing or
decreasing the Receive Gain (Rx) of the Mobile port. By default Rx is set at 3.
If you change the Tx or Rx Gain during an active call, the change you made will not apply on the current
call. It will be applied on the next call.
• Rx Gain at SIP Trunk/SIP Extension (dB): This parameter allows you to increase the incoming speech
volume level of calls from SIP trunks/Extensions to Mobile Ports. By default, Rx Gain is set to 0dB.
• Tx Gain at SIP Trunk/SIP Extension (dB): This parameter allows you to increase the outgoing speech
volume level of calls from SIP trunks/Extensions to Mobile Ports. By default, Tx Gain is set to 6dB.
You can set Rx and Tx Gain at SIP Trunks/Extensions using SE Commands only. For instructions, see
“Configuring Mobile Port Parameters using a Telephone” later in this topic.
• Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this mobile trunk port:
• Type: Select the type of Call Budget on Trunk—Amount, Minutes or Number of Calls—to be applied on
this mobile trunk port. By default, no Call Budget type is selected.
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
• Calls: If you selected ‘Calls’ as the Call Budget Type, enter the number of calls in this field. By default,
the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
• Scheduled (Date): Enter the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select ‘Daily’ if your plan suggests
so.
The consumed Call Budget can be reset from SE and SA Mode, referred to as Manual Reset. Refer the
feature description “Call Budget on Trunk”.
• Call Back: This parameter is related to the ‘Call Back on Trunk Port’ feature. If you want to enable the 'Call
Back on Trunk Port' feature on this Mobile trunk, configure the following parameters:
86. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-programma-
ble.
• Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote party
should be routed:
• DID
• PIN Authentication - Multiple Calls
• CLI Authentication - Multiple Calls
• CLI Authentication - Single Call - Answer Signaling
• Operator
• Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the ‘CLI number”. If you want the call back to be made to a different number, select ‘Alternate
Number’.
• Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all Mobile trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all Mobile trunks and Trunk Port types, retain
this list.
If you want a different set of number strings to be programmed for this Mobile Trunk, select a different
Number List, and assign it to the Mobile trunk port.
You may program the Incoming Number List either from the ‘Number List’ page or by clicking the
‘Incoming Number List’ link to reach the Number List page.
Refer the topic “Number List” to know more, and for configuration instructions.
• Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the ‘Incoming Number List’, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the ‘Outgoing Number List’.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all Mobile trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this Mobile Trunk
port.
You may program the Outgoing Number List either from the ‘Number List’ page or by clicking the
‘Outgoing Number List’ link to reach the Number List page.
• Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an Outgoing Trunk Bundle Group (OTGTBG).
Select ‘Same port’ if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
• OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
• If you want the system to select the lowest cost trunk for making the call back, enable Least Cost Routing
on the OGTBG that you define here for Call Back.
• Accept Anonymous Calls: The flag is for accepting calls without CLI that land on the mobile port. By
default the flag is enabled. You may disable this flag to disallow calls without CLI on this Mobile port.
• Pause Timer: This Timer is used for providing delay in number dialing from the Mobile port. The Pause
Timer will be applicable when the digit 'P' is configured in the DTMF number string which is to be out dialed
as DTMF digits on the Mobile port.
For example, if PPP2 is to be out dialed and Pause timer is programmed as 3 seconds, the ETERNITY will
out dial the digit 2 after 9 seconds, that is, after a delay of individual P (3+3+3 =9). The range of this time is
from 1 to 9. By default the Timer is set to 3 seconds.
• DTMF Outdial: You can select whether to send the DTMF digits from the Mobile Ports In-band or through
signaling, that is, AT Command. Default: AT Command.
When you select DTMF Outdial using AT Command, the length of the DTMF digits will be determined by
the DTMF ON Time you set.
• DTMF ON Time: This parameter determines the time for which the DTMF digit will remain ON, while being
out dialed by the ETERNITY. This parameter finds its application in the feature “Multi-Stage Dialing”' and in
DTMF Outdialing using AT Command.
• DTMF Detection Minimum Level (dB): Define the minimum dB level for detecting the DTMF digits dialed.
Default: -30
• Category (Logical Partition): This parameter assigns the Mobile Port to a trunk category for the purpose
of Logical Partitioning. By default all Mobile Ports are assigned to Category 1. Do not change the default
setting.
If you want to change the call permission between the mobile port and other trunks, click the 'Category' link
to open the Logical Partitioning page. You may program the call permission between Category 1 (assigned
to Mobile Trunk Ports) and other Categories. Refer the feature description “Logical Partition” to know
more.
• Use?: Enable this flag if you want the Mobile port to be used in a Gateway Application.
• DTMF String (max. 4 digits): Program the DTMF digits to be sent to signal call maturity to the source
port.
• Debug: Enable this flag by selecting the check box if you want to initiate debugging for the Mobile Port. By
default, debugging is disabled.
• If you have completed configuration of all the above listed Mobile Port Parameters, click 'Submit' at the
bottom of the page to save your changes.
• For Advanced Configuration of the Mobile Ports, use the following commands:
• For commands to program Call Budget on Mobile Ports, refer the topic “Call Budget on Trunk”.
To program the DTMF String for the Gateway Application on the Mobile Port, dial:
• 8017-1-Mobile-DTMF String to program the string on a single mobile port.
• 8017-2-Mobile-Mobile-DTMF String to program the same string on a range of mobile ports.
• 8017-*-DTMF String to program the same string on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
DTMF String is a maximum of 4 digits. Default: CCC
Network Selection
After the Mobile card is successfully installed and powered on, the mobile port is programmed to automatically
locate and register with the Network that supports the SIM card installed in. Also, at each power ON, the mobile
port (SIM) will automatically register with the Network that supports the SIM on it.
However, if the Mobile port fails to register, it will restart the process of network selection on the expiry of the
Network Registration Retry Timer87.
If the ETERNITY is located in a border area where more than one Network Operator is available, it is possible that
the SIM card may register with another available network and result in 'Roaming' charges. To avoid this, you must
disable automatic network selection and program manual network selection.
87. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-programma-
ble.
If no match is found, the Mobile port (SIM) will not get registered with any of the available network operators and no
calls can be made or received on this port.
• Click 'Network Selection Mode' of the desired Mobile port. Select 'Manual'.
• Enter the Network Operator Codes (MCC-MNC) in order of priority. The codes must not exceed 8 digits.
You can store up to 9 Network Operator Codes in the order of priority.
• Repeat the same steps to set network selection mode for other mobile ports.
• When you change the Network Selection Mode to ‘Manual’ and the Network Operator Code manually,
the change you made will not come into effect until you have restarted the Mobile Port.
88. The Network Operator Code comprises of the Mobile Country Code (MCC) appended by the Mobile Network Code (MNC). The
MCC is usually a 3-digit code that identifies a country. A single country may be assigned more than one MCC. For example the
MCC assigned to India is 404, but same code applies to all network operators in the country.
The MNC is usually a 2/3-digit code. The MCC-MNC combination uniquely identifies the home network of the mobile terminal or
the mobile user. For example, AirTel, a GSM network operator in India, has different MNC assigned to its networks in various
states. The MNC for AirTel in the state of Maharashtra is 90, while the same for the state of Gujarat is 98.
For example, to program Manual Network Selection on Mobile Port number 1, dial 8007-1-01-2
To program the network operator codes in order of priority for Mobile port, dial:
• 8008-1-Mobile-Priority-Network Operator Code-#* to program network operator codes for a single
mobile port.
• 8008-2-Mobile-Mobile-Priority-Network Operator Code-#* to program the same network operator
codes for a range of mobile ports.
• 8008-*-Priority-Network Operator Code-#* to program the same network operator codes of all mobile
ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
Priority is from 1 to 9.
Network Operator Code is MCC-NCC maximum 8 digits. Terminate the command with #* if the number
of digits is fewer than 8.
For example, following Network Operator Codes need to be programmed in the order of priority on
Mobile Port number 01:
Priority 1=40498
Priority 2 = 40425
Priority 3 = 40421
Dial:
8008-1-01-1-40498-#*
8008-1-01-2-40425-#*
8008-1-01-3-40421-#*
• Exit SE mode.
• Wait for 4-5 seconds after the Port Status page is opened.
• The Port status page will display the following parameters for all Mobile ports that have been enabled:
• Port Name: This is the name by which the Mobile Port is programmed.
• Port Status: This is the status of the connection - showing Initialization with the Network, Registering
with the Network, Idle or Busy state of the network. It also shows errors and alerts when SIM is absent,
the wrong SIM PIN has been entered, SIM PUK is required.
• IMEI: This is the unique identification number of (the GSM engine) each Mobile port.
• Network Operator Name: This is the name of the service provider/network operator with which the
Mobile Port is registered.
• Signal Strength (dBm): This is the signal strength in '-dBm' as received from the network with which
the Mobile port is registered.
• Bit Error Rate (BER): BER is Bit Error Rate which defines the quality of the channel.
• Call Duration: This is the total call duration of matured outgoing calls89 on the Mobile port. This data is
used for calculating Answer Seizure Ratio (ASR) for the port. It is displayed in MMMMMM:SS format.
• Dialed Calls: This is the total number of outgoing calls90 made from the Mobile port. This data is used
for calculating Answer Seizure Ratio (ASR) for the port.
• Successful Calls: This is the total number of matured outgoing calls made from the Mobile port. This
data is used for calculating Answer Seizure Ratio (ASR) and Average Call Duration for the port.
• ASR: This is the Answer Seizure Ratio (ASR) calculated by the system for the Mobile port, in terms of
percentage. ASR is the sum of all outgoing matured calls from the Mobile port, divided by the total
number of outgoing calls made from the Mobile port, multiplied by 100. The system calculates ASR
after the completion of the outgoing call.
• ACD: This is the Average Call Duration (ACD) of outgoing calls made from the Mobile port. It is an
indicator for monitoring the network condition. Decreasing ACD is indicative of trouble in the network
condition.
The system calculates ACD after the completion of the outgoing calls, by dividing the total call duration
by the number of outgoing matured calls.
• Reset ASR and ACD: This field allows the System Engineer to reset manually the ASR and the ACD
of the Mobile port.
The parameters Total Call duration, Number of matured calls, Total Number of OG Calls, ASR and
ACD are saved in the configuration, and are not reset on Power OFF condition. The system maintains
the statistics for the last 999 calls. When the total number of outgoing calls exceeds 999, the system
will stop calculating ACD and ASR and will display ASR and ACD calculated on the basis of the last
999 calls only.
Therefore, the System Engineer must manually reset ASR and ACD when the total number of calls
reaches 999. When you reset ASR and ACD the number of call matured and the number of calls dialled
is reset to 0.
ASR and ACD can be reset anytime, even when the total number of calls is less than 999.
When ACD is reset, only the 'Total Call Duration' maintained for the ACD calculation will be reset. The
'Total Call Duration' of the Call Budget, the consumed minutes maintained for the Call Budget on the
mobile port will remain unaffected.
89. Matured calls are outgoing calls for which 'CONNECT' message was received from the network.
90. The total number of outgoing calls made includes the number of times the ATD has been sent from the Mobile port to the network.
• IMSI: International Mobile Subscriber Identity (IMSI) is a unique number stored in the SIM card.
• Cell ID: This is the 16-bit identifier that identifies the cell. The cell is the radio coverage area given by
one BTS (Base Transceiver Station).
• Location Area Code (LAC): The LA (Location Area) is a group of cells defined by the Operator. The
LAC (Location Area Code) uniquely identifies a LA within a PLMN (Public Land Mobile Network).
• Call Budget Type: This shows the Call Budget Type, whether Amount, Minutes or Number of Calls,
are set on the Mobile port.
• Allotted Amount (Rs.) / Minutes/Calls: This shows the sum/number of minutes/number of calls
allotted as Call Budget on the Mobile port.
• Consumed Amount (Rs.) / Minutes/Calls: This shows the sum/number of minutes/number of calls of
the allotted Call Budget that has been used up on the Mobile port.
• Call Budget Reset Mode: This shows the whether manual or scheduled reset of the consumed call
budget is set on the Mobile port.
• Call Budget Reset Schedule (Date): This shows whether the consumed Call Budget on the Mobile
port is to be reset Daily or on a particular date of a month.
• Reset Consumed Amount/Minutes/Calls: This editable field allows the System Engineer to reset the
consumed Call Budget Amount/Minutes/Calls at any time, manually.
• Registered with Network: This shows the type of network with which the Mobile port is registered,
whether GSM, GSM Compact, 3G or UMTS.
• Firmware Version of Engine: This shows the firmware version of the mobile Engine.
• Exit SE mode.
• view the ID of the Mobile Network Operator with which the Mobile Port is currently registered.
• check Signal Strength of a mobile port, whenever there is trouble placing calls over a Mobile port to rule
out weak signal as the cause.
When the network responds, you the Mobile Network Signal Strength will be displayed on the LCD of
your DKP and programming beeps will be played.
The Signal Strength values are in -dBm. '-113' indicates weak signal, whereas '-51' indicates maximum
signal strength.
You will get the confirmation tone and the confirmation message “ASR-ACD Reset” will appear on your
phone’s display.
After you have installed the VoIP Card and checked its functioning, first program the VoIP Port
Parameters, followed by the SIP Trunk Parameters, and SIP Extensions, as required.
The VoIP Card may be installed typically, in a Public IP Network or in a Private network, behind a NAT Router.
When the VoIP Card is installed in a Private Network, behind a NAT Router,
• the WAN Port of the card is connected to the LAN Switch/Hub.
• Private IP is assigned to the WAN Port.
• SIP devices within the LAN can get registered with the card.
When your VoIP Card is installed in a Private Network, you may have to change the IP Address and
Subnet Mask of the WAN Port of the card, before connecting it to the LAN Switch/Hub. However, this will
not be necessary, if there is a DHCP server on the LAN which will automatically assign an IP Address that
does not conflict with any other device on the LAN.
Depending on your installation scenario, configure the VoIP Port Parameters using Jeeves or dialing commands
from a Telephone.
• Hardware Slot: This is the number of the Universal Slot in which the VoIP card has been inserted. The
ETERNITY can automatically detect and assign the Hardware Slot numbers to the VoIP card (LAN and
WAN Port). If required, you may change the Hardware Slot number.
• Name: You may assign a 'Name' to each VoIP Port to facilitate easy identification. The Name may
comprise a maximum of 18 characters.
LAN Port
• MAC Address: This non-editable field shows the MAC Address of the LAN port.
• IP Address: Enter the IP Address to be assigned to the LAN Port of the VoIP Card. The default IP
Address is 192.168.002.031. You can assign only Static IP to the LAN Port.
• Subnet Mask: Enter the Subnet Mask to be assigned to the LAN Port. The default Subnet Mask is
255.255.255.0
WAN Port
• MAC Address: This non-editable field displays the MAC Address of the WAN port.
• Use MAC Cloning: MAC Cloning is required when you want the WAN Port to use a MAC Address other
than its own unique MAC Address as source MAC Address.
Select the check-box to enable cloning of the MAC Address of the WAN Port. By default, MAC Address
Cloning is disabled.
• Clone MAC Address: If you have enabled MAC Cloning, enter the MAC Address to be cloned in this field.
• Connection Type: Select the appropriate Connection Type for the WAN port, according to the IP
Addressing scheme of your installation scenario. Consult your LAN Administrator also in this regard.
• Static: Select this option if the connection type is Static. This is also the default connection type for all
WAN ports.
• DHCP: Select this option if the connection type DHCP. As the DHCP Server will automatically assign IP
Address, Subnet Mask, Gateway Address to the WAN Port, you need not configure any of these.
• PPPoE: Select this option if the connection type is PPPoE. As the PPPoE server will automatically
assign the IP Address, Subnet Mask and Gateway Address to the WAN Port, you need not change any
of these. You must program the User ID, Password and PPPoE Service Name as provided by your ISP.
Program the Service Name only if it has been provided. You must set DNS address.
• PPPoE: The parameter is relevant if you have selected PPPoE as the Connection Type. Configure the
following PPPoE parameters:
• User ID: Enter the User ID provided by the Internet Service Provider. The User ID may be a maximum
of 16 characters.
• Password: Enter the User Password provided by the Internet Service Provider. The password may be
a maximum of 16 characters.
• Service Name: Enter the PPPoE Service Name, if provided by your Internet Service Provider. The
Service Name may consist of a maximum of 16 characters. If Service Name is not required, leave this
field blank.
• IP Address: You must enter the IP Address, only if you selected 'Static' as the Connection Type.
If you selected DHCP or PPPoE as the Connection Type, the IP Address assigned by the DHCP/PPPoE
server will be displayed here.
• Subnet Mask: You must enter the Subnet Mask, only if you selected 'Static' as the Connection Type.
If you selected DHCP/PPPoE as the Connection Type, the Subnet Mask assigned by the DHCP/PPPoE
server will be displayed here.
If you selected DHCP/PPPoE as the Connection Type, the Gateway IP Address assigned by the DHCP/
PPPoE server will appear here in non-editable format.
• Domain Name Server (DNS): Configure the following DNS Connection settings for the WAN Port:
• DNS Address Assignment: If you selected 'Static' as your network Connection Type (IP Addressing),
you can select only 'Static' as the DNS Address Assignment.
If you selected DHCP as your network Connection Type, and the DHCP server provides DNS Address,
set the DNS Address Assignment to 'Automatic'. If the DHCP server does not provide DNS Address,
set DNS Address Assignment as 'Static' and program the DNS Server Address/DNS Name provided by
your ISP.
If you selected PPPoE as your network Connection Type, and the PPPoE server provides DNS
Address, set the DNS Address Assignment as 'Automatic'. If the PPPoE server does not provide DNS
Address, set the DNS Address Assignment as 'Static' and program the DNS Server Address/DNS
Name provided by your ISP.
• DNS Address: This field will be editable only if you selected DNS Address Assignment as 'Static'.
Enter the DNS IP Address here. The DNS Address can be a maximum of 15 characters.
If you selected DNS Address Assignment as 'Automatic', the DNS Address assigned by the DHCP/
PPPoE server will appear here.
• DNS Domain Name: Program DNS Domain Name if provided by your ITSP/LAN Administrator.
Otherwise, keep this field blank. The Domain Name may be a maximum of 40 characters.
• Dynamic DNS (DynDNS.org): This parameter is applicable only when you are going to configure SIP
Extensions on the VoIP Card.
When the VoIP Card is assigned dynamic IP Address using DHCP or PPPoE, SIP-enabled devices
registered with the Card as SIP Extensions need to change their configuration whenever a new IP Address
is assigned to the VoIP Card. Dynamic DNS resolves this.
ETERNITY VoIP Card supports Dynamic DNS Server client of the Service Provider Dynamic DNS.org. If
you want to use the DNS Service of DynDNS.org, program these parameters:
• Enable Dynamic DNS?: If you have taken the services of DynDNS.org, you must enable this flag. By
default this flag is disabled.
• Update IP Address at Power ON?: When your VoIP Card is registered with the DynDNS.org, the
DynamicDNS server stores the mapping between hostname and IP Address, which can be updated
periodically. However, if the VoIP card frequently sends IP Address update request to the DDNS server,
the server is likely to block the hostname in its database and terminate the DDNS services provided to
you.
So, if you restart the ETERNITY frequently, there is a great chance that DDNS server will block the
hostname programmed in the system. This will in turn affect the ability of the system to receive the calls
The VoIP Card offers you control over whether the system should update the IP Address in the DDNS
server at each Power ON or not. It also allows you to update the IP Address at any time, as required.
If you do not want to update the IP Address in the DDNS Server at each Power ON, set this flag to 'No'.
By default, the flag is set to 'No'.
• User ID: Enter the User ID created by you with DynDNS.org here. A maximum of 40 characters,
including all ASCII characters are allowed.
• Password: Enter the Password created by you for your User ID with DynDNS.org here. The password
may be not more than 24-characters long.
• Host Name: Enter the Host Name created by you with DynDNS.org here. A maximum of 40
characters, including all ASCII characters are allowed as Host Name.
• Retry Trials: This count defines the number of attempts that the VoIP Card should make to send the IP
Address Update Request to the Dynamic DNS Server. The Retry Count may be set from 1 to 9. By
default the count is set to 1.
• Update IP Address Now?: Enable this flag whenever you want to update the IP Address in the DDNS
server. By default, this flag is disabled.
You can use this flag to update the IP Address in the DDNS server, if you have disabled Update IP
Address during each Power ON.
• VoIP Server Domain: This parameter is of relevance only if you are configuring SIP Extensions on the
VoIP Card.
The VoIP Card is capable of maintaining a domain for registering SIP clients (any SIP-enabled device) as
SIP Extensions.
Program the Server Domain if you want SIP clients to register with the Registrar Server of the VoIP card
using the domain handled by the VoIP Card91.
If you program Server Domain for registration of SIP clients, you must also map the Domain name and the
IP Address of the WAN Port of the VoIP Card to the DNS Server in the network.
91. SIP clients can be registered with the VoIP Card either using the domain handled by the VoIP Card or using the WAN or LAN Port
IP Address.
If domain is programmed, VoIP card will listen for the SIP message which is redirected to the programmed domain only. It will also
listen for SIP messages on the WAN IP address and LAN IP address.
But if domain is not programmed, the VoIP card will listen for SIP messages only on the WAN IP Address and LAN IP address.
• Quality of Service (QoS): This refers to priority of IP packets on network layer. QoS is programmed for
both signaling (SIP) and media (RTP). following types of QoS can be configured:
• SIP DiffServe/ToS: The VoIP Card sends all the SIP signaling messages with this QoS setting which
is selected here. This field defines the priority bits for SIP messages. There are two types of SIP QoS
from which you can select.
These are: Precedence and DiffServe. By default DiffServe is selected. Set the range of the selected
SIP QoS.
• RTP DiffServe/ToS: The VoIP Card sends all the RTP packets with the QoS setting which is selected
here. This field defines the priority bits for RTP packet. There are two types of RTP QoS from which
you can select for RTP: Precedence and DiffServe. By default DiffServe is selected. Set the range of
the selected RTP QoS.
QoS parameters are applicable for all packets (SIP/ RTP) leaving both LAN and WAN port as well as TCP
connection.
• Simple Traversal of UDPs through NATs (STUN):This parameter is to be configured only if the VoIP
Port (WAN) ETERNITY is located behind a NAT Router and SIP Messages need to be forwarded to the
public internet.
Simple Traversal of UDP through NAT (STUN) specifies the mechanism required for NAT traversal in SIP
messages. The STUN Server facilitates traversing through most NATs, except symmetric NATs. If your
router has symmetric NAT, do not program this parameter. If your router as asymmetric NAT, configure the
following STUN parameters:
• STUN Server Address: Enter the STUN Server Address, a maximum of 40 characters.
• STUN Port: Enter the Listening Port of the STUN Server. The valid range for this field is from 1024-
65535. The default STUN Port is 03478.
• Use SIP Port fetched using STUN: This flag is enabled by default to allow SIP Port Number to be
fetched using STUN in the SIP message. Disable this flag if you are using Port-Forwarding in the
Router for SIP messages.
Since STUN does not work with symmetric NAT, as an alternative to STUN you can use the Router's Public
IP Address as NAT Traversal mechanism. Ask your Network Administrator about the NAT Traversal
mechanism that suits best for your voice network and program this parameter.
• Router's Public IP Address: This parameter is of relevance if the VoIP Port (WAN) of the ETERNITY is
located behind a NAT Router and SIP Messages are to be forwarded to the public internet.
Router's Public IP Address specifies the fixed IP Address of your NAT router required for NAT Traversal in
SIP messages.
You also need to select 'Router's Public IP Address' as the 'Source Port IP Address' in the 'SIP Extension
General Parameters', when you configure SIP Extensions'. Refer the topic "Configuring SIP Extensions".
You can also use STUN as an alternative to the Router's Public IP Address as NAT Traversal mechanism.
Ask your Network Administrator about the NAT Traversal mechanism that suits best for your voice network
and program this parameter.
• Disconnect on Silence Detection: With this parameter, any matured incoming or outgoing call can be
disconnected automatically, if silence (No RTP Packets) is detected for more than a specified duration of
time.
• Enable Disconnection on Silence Detection?: If you want to use this feature, click the check box to
enable it. By default, Disconnect on Silence Detection is enabled.
• Detection Time (sec): This Timer defines the duration for which, if silence is detected continuously,
the call will be disconnected. The valid range of this Timer is from 001 to 999 seconds. By default, it is
set to 999 seconds.
• Channels Reserved for SIP Trunks: The VoIP Card supports up to 32 voice channels (depending on the
model of ETERNITY), which can be used by SIP Extensions and SIP trunks.
It may happen that SIP Extension users use up most of the channels of the VoIP card, leaving too few or
none for making/receiving SIP Trunk calls.
This can be avoided by reserving some voice channels exclusively for SIP trunk calls.
Specify the minimum number of voice channels you want to reserve for SIP Trunk calls. By default, no
channel is reserved.
• SIP 100rel: This parameter is to be configured if you want to support reliable transmission of (SIP)
provisional responses. Enable 100rel by selecting the check box, if you want the VoIP Port to use 100rel
for reliable transmission of SIP provisional responses and to use PRACK (Provisional Acknowledgement).
By default, the flag is disabled.
Enable this flag if you want to receive SIP messages over TCP. To be able to send SIP messages over
TCP, you must configure 'TCP' or 'TCP (Fallback to UDP)'. By default, SIP Over TCP is enabled.
• SIP UDP Port: This port defines the port on which the VoIP Port of ETERNITY listens for SIP messages
transported over UDP. This port is also used as the source port for sending SIP messages to the remote
peer. The valid range for this port is 1024-65535. Default: 05060.
• SIP TCP Port: This port defines the port on which the VoIP Port of ETERNITY listens for SIP messages
transported over TCP. This port is also used as the source port for sending SIP messages to the remote
peer. The valid range for this port is 1025-65535. The default SIP TCP Port is 05060.
• RTP Listening Port: This port defines the port on which the VoIP Port of ETERNITY listens for RTP
Packets. This port is also used as the source port for sending RTP packets to the remote peer. The valid
range for this port is 1024-65278. The default RTP Listening Port is 08000.
• Layer 2 VLAN/CoS: This parameter is to be configured if the VoIP Port (WAN) of ETERNITY is to be
connected in VLAN network.
This parameter enables the ETERNITY to add VLAN header to the packets generated by it. The VLAN
header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of
traffic92.
VLAN Tag is applied on all packets generated by system (SIP, RTP, DNS, ARP, etc.), whereas CoS bits
are applied only for SIP and RTP packets generated by system.
The corresponding meaning of CoS bits with respect to traffic type is as follows:
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
• Enable?: When this flag is enabled, all packets generated by the system (SIP, RTP, DNS, ARP, etc.)
will be tagged with VLAN ID as programmed. The CoS bits as programmed for SIP and RTP packets
will be included in the VLAN header. By default, this flag is disabled.
92. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), better handling
of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred to
as Class of Service (CoS) which is defined by IEEE 802.1P.
• SIP CoS: Define the CoS (priority) bits in all SIP packets. The range of CoS bits is from 0 to 7. Default:
3.
• RTP CoS: Define the CoS (priority) bits in all RTP packets. The range of CoS bits is from 0 to 7.
Default: 6.
• UDP NAT Keep Alive: This parameter is to be configured when the VoIP Port is connected behind a NAT
router93 and SIP messages are transported over UDP. UDP NAT Keep Alive messages must be sent to
refresh the UDP binding in the NAT router.
• Enable: Enable this flag to send UDP NAT Keep Alive messages periodically to refresh the binding in
the NAT router. By default, this flag is disabled.
• Interval (sec): Select Time period after which the VoIP Port should send UDP NAT Keep Alive
messages. This time period should be less than the UDP Binding Timer of the router. The valid range is
001-999 seconds. By default it is set to 120 seconds.
• Type of Message: Select the type of message type to be sent when UDP NAT Keep Alive is enabled.
Select either REGISTER or NOTIFY. By default, NOTIFY is selected.
• TCP NAT Keep Alive: This parameter is to configured when the VoIP (WAN) Port is connected behind a
NAT router and SIP messages are transported over TCP. TCP NAT Keep Alive messages must be sent to
refresh the TCP binding in the NAT router.
• Enable: Enable this flag to send TCP NAT Keep Alive messages periodically to refresh the binding in
the NAT router. By default, this flag is disabled.
• Interval: Select Time period after which the VoIP Port should send TCP NAT Keep Alive messages.
This time period should be less than the Binding Timer of the router. The valid range is 001-999
seconds. By default it is set to 120 seconds.
• SIP Invite Timer (sec): This is the time in seconds that the VoIP Port waits for a response from the called
party after ending INVITE message. This timer starts after sending INVITE message to the called party
and stops on receipt of the provisional response or the final response or when the user disconnects the
call. On expiry of the timer, the call process is terminated by the ETERNITY and an error tone is played to
the user. The range of the SIP Invite Timer is 010-180 seconds. By default it is set to 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds that the VoIP Port waits for the final response
after receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called party
or when the user disconnects the call. On expiry of the timer, the ETERNITY terminates the call process
and plays error tone to the user. The range of SIP provisional Timer is 010-180. By default, the timer is set
to 60 seconds.
• General Request Timer (sec): The time in seconds for which the VoIP Port waits for response for a
transaction request. This timer starts on the initiation of a transaction. This timer stops on receipt of a
93. Network Address Traversal (NAT) allows multiple hosts in the network to share the single public routable IP address. Means all the
hosts in the private network shall be identified by single public IP address in the global IP cloud.
• Assign LED for SIP Trunk: You can monitor the status and functioning of any one of the SIP trunks with
this parameter. This is LED2 on the VoIP card, which is assigned to any one of the SIP trunks that you
want to monitor. When assigned to a SIP trunk, the LED indicates the functioning of this trunk. By default
the LED is assigned to SIP trunk 01. Enter the number of the SIP trunk you want to be monitored here.
• If you have completed configuring the required parameters for the VoIP Ports, click 'Submit' at the bottom
of the page to save your VoIP Port settings.
• To program the parameters or another port, click the VoIP Port number on the top of the screen. Repeat
the same steps as above. Click ‘Submit’ at the bottom of the page to save your port parameter settings.
Repeat this command to program the Cloned MAC Address on another VoIP Port.
When you enter the number string of the Cloned MAC Address, alphanumeric dialing will be automatically
enabled on the DKP. Do not enter '.' (dot/period) in the MAC Address. For example, to program MAC
Address 00.50.C2.55.B0.10, enter 0050C255B010 from the DKP.
To prevent the possible collision and loss of data, do not clone the MAC address, which is already used by
another device in the same network. Also, use MAC address as per IANA (Internet Assigned Numbers
Authority) standards.
To configure IP Address for VoIP Port having 'Static' as Connection Type, dial:
• 7754-1-VoIP Port-IP Address
Where,
VoIP Port is the number of the software port from 01 to 16
IP Address is from 000-255.
Enter each octet in full. For example: To program IP address 192.168.10.11, enter 192168010011.
To configure Subnet Mask for the VoIP Port having 'Static' as Connection Type, dial:
• 7755-1-VoIP Port-Subnet Mask to program the Subnet of a single VoIP port.
• 7755-2-VoIP Port-VoIP Port-Subnet Mask to program the same Subnet for a range of VoIP ports.
• 7755-*-Subnet Mask to program the same Subnet for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
Subnet Mask is from 000-255. Enter each octet in full. For example to program Subnet Mask
255.255.255.0, enter 255255255000.
To configure Default Gateway IP Address for the VoIP Port having 'Static' as Connection Type, dial:
• 7756-1-VoIP Port-Gateway Address to program Gateway IP Address for a single VoIP port.
• 7756-2-VoIP Port-VoIP Port-Gateway Address to program the same Gateway IP Address for a range
of VoIP ports.
• 7756-*-Gateway Address to program Gateway IP Address for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
Gateway IP Address is from 000-255. Enter each octet in full. For example: Enter the Gateway Address
192.168.10.10 as 192168010010.
Dynamic DNS
SIP 100rel
To enable/disable Use 100rel Response for a VoIP Port, dial:
• 7783-1-VoIP Port-Code to enable/disable 100rel Response for a single VoIP port.
• 7783-2-VoIP Port-VoIP Port Code to enable/disable 100rel Response for a range of VoIP ports.
• 7783-*-Code to enable/disable 100rel Response for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Code is
0 for Disable
1 for Enable
Default: Disable
Layer 2 VLAN/CoS
To enable/disable Layer2 VLAN/CoS for a VoIP Port, dial:
• 7775-1-VoIP Port-Flag to enable/disable Layer2 VLAN/CoS for a single VoIP port.
• 7775-2-VoIP Port-VoIP Port-Flag to enable/disable Layer2 VLAN/CoS for a range of VoIP ports.
• 7775-*-Flag to enable/disable Layer2 VLAN/CoS for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Flag is
0 for Disable
1 for Enable
Default: Disable
To define Interval for UDP NAT Keep Alive for a VoIP Port, dial:
• 7762-1-VoIP Port-UDP NAT Keep Alive Interval to define the interval for a single VoIP port.
• 7762-2-VoIP Port-VoIP Port-UDP NAT Keep Alive Interval to define the same interval for a range of
VoIP ports.
To select the Type of Message for UDP NAT Keep Alive for a VoIP Port, dial:
• 7778-1-VoIP Port-NAT Keep Alive Message Type to select message type on a single VoIP port.
• 7778-2-VoIP Port-VoIP Port-NAT Keep Alive Message Type to select the same message type for a
range of VoIP ports.
• 7778-*-NAT Keep Alive Message Type to select the same message type on all ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
NAT Keep Alive Message is
1 for NOTIFY
2 for REGISTER
Default: NOTIFY
To define Interval for TCP NAT Keep Alive for a VoIP Port, dial:
• 7786-1-VoIP Port-NAT Keep Alive Interval for TCP to define interval for a single VoIP port.
• 7786-2-VoIP Port-VoIP Port-NAT Keep Alive Interval for TCP to define the same interval for a range
of VoIP ports.
• 7786-*-NAT Keep Alive Interval for TCP to define the same interval for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
NAT Keep Alive Interval for TCP is from 001 to 999 seconds.
Default: 120 seconds
Timers
To configure the SIP Invite Timer, dial:
• 7770-1-VoIP Port-SIP Invite Timer to configure the invite timer for a single VoIP port.
• 770-2-VoIP Port -VoIP Port-SIP Invite Timer to configure the same timer duration for a range of VoIP
ports.
• 7770-*-SIP Invite Timer to configure the same timer duration for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP Invite Timer is from 010 to 180 seconds.
Default: 30 seconds.
To configure the SIP General Request Timer for a VoIP Port, dial:
• 7779-1-VoIP Port-SIP General Request Timer to configure the request timer for a single VoIP port.
• 7779-2-VoIP Port-VoIP Port-SIP General Request Timer to configure the same timer duration for a
range of VoIP ports.
• 7779-*-SIP General Request Timer to configure the same timer duration for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP General Request Timer is from 10 to 60 seconds.
Default: 20 seconds.
LED Indication
To assign Registration LED to a SIP trunk:
• 7772-1-VoIP Port-SIP trunk
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP trunk is the number of the SIP trunk you want to assign the LED, from 01 to 32.
Default: SIP trunk 01.
The Port status page will display the following parameters for all VoIP Ports you have configured:
SIP trunks may be Proxy or Non-Proxy. All SIP trunks are considered as Non-Proxy trunk by default.
Regardless of whether a SIP Trunk is Proxy or Non-Proxy, it must be assigned to a VoIP Ethernet Port, which is to
be used for that SIP trunk and the SIP trunk must be enabled. VoIP Calls can be initiated after suitable
configuration of the SIP trunk number in the Outgoing Trunk Bundle Group.
For Proxy SIP Trunks, you must program the following parameters required for registration with the Proxy Server.
• Program the SIP ID, registrar Server Address, Registrar Server Port, Authentication ID, Authentication
Password as provided by your ITSP.
• If your ITSP uses Outbound Proxy, Enable the Outbound Proxy for the SIP trunk and also program the
Outbound Proxy Server Address and Port as provided by your ITSP.
The VoIP Ethernet Ports are assigned Software Port numbers from 01 to 16. Configuration of all VoIP
parameters are done for the Software Ports. Each of these software ports must first be assigned a Hardware
ID. A hardware ID of the Software Port shows the physical location of the port on the system.
Please read “Software Port and Hardware ID” to know about know more about this topic.
If you have installed as single VoIP card, you can configure all SIP trunks on the same card.
If you have installed more than one VoIP card, you may program the SIP trunks on the VoIP Ethernet Ports in
a distributed manner. For example: if you have installed four VoIP cards in your ETERNITY ME, you can either
program 8 Trunks each on the 4 VoIP Ports OR you may program 12 Trunks on the VoIP Port of the first card,
8 on the VoIP Port of the second card, 6 Trunks each on the VoIP Ports of the third and the fourth card.
• Scroll to 'SIP Trunk Parameters' and click this link to open the page.
• SIP Trunk No.: This non-editable field is the number of the SIP trunk. The SIP trunks are numbered
from 01 to 32.
• VoIP Ethernet Port No.: This is the software port number of the VoIP Ethernet Port, from 00 to 16, to
which you want to configure the SIP trunk.
It is possible to configure more than one SIP trunk on a single VoIP Ethernet Port. To do this, assign
the same VoIP Ethernet Port Number to all those SIP trunks you want to configure on that VoIP Port.
For example, if you want to configure SIP trunks 03, 04 and 05 on VoIP Ethernet Port No. 02, enter this
number (02) in this field for SIP trunk number 03, 04, and 05.
• Enable SIP Trunk: This flag is for enabling or disabling the SIP trunk. To be able to make incoming
and outgoing calls from the SIP trunk, click to enable the SIP trunk. By default, the SIP trunk is
disabled, disallowing incoming and outgoing calls. You may disable SIP trunks that are not functioning.
The Name assigned to the SIP trunk may consist of a maximum of 18 characters. The Name of the port
may be the name of the ITSP the SIP trunk is subscribed with (recommended).
• SIP User ID: Enter the SIP User ID provided by the ITSP. SIP User ID is the ID that callers will use to
call this SIP trunk.
The SIP User ID may be a number or text for remote parties to call on the SIP trunk. For example, if
SIP URI provided by the ITSP is [email protected], enter 12345 in this field. SIP User ID may consist of
a maximum of 40 characters. All ASCII characters are allowed.
• Registrar Server Address: Enter the Proxy/Registrar Server Address. The Server Address may be an
IP Address or a Domain name, of maximum 40 characters.
• Registrar Server Port: Enter the Registrar Server Listening Port. The valid rang is from 1024 to
65535. By default, 5060 is set as the Listening Port of the Registrar Server.
• Re-registration Time (sec): The Registrar Server deletes an entry of its client from its database on
expiry of a fixed timer, which is set by the Registrar Server. ETERNITY VoIP Card sends a registration
request before this Timer expires to remain registered on the server.
Enter the value of the Timer after which the system should send registration request to maintain
registration binding with the server. The valid range of this timer is from 00001- 65535. By default the
Timer is set to 3600 seconds.
• Registration Retry Time (sec): This Timer stands for the period between retries for registration. If the
registration attempt fails, ETERNITY sends the registration request on the expiry of this Timer again.
The system continues to send the registration request till it gets registered. The valid range of this timer
is from 00001- 65535. By default the Timer is set to 00010 seconds.
• Authentication User ID: Enter the Authentication ID provided by the ITSP for this SIP trunk. The
Authentication User ID may be a string of 40 characters (maximum), including ASCII characters.
• Authentication Password: Enter the Authentication Password provided by the ITSP for this SIP trunk.
The Authentication Password may be a string of 16 characters (maximum), including ASCII characters.
• Outbound Proxy: This parameter is relevant only if the ITSP has a SIP outbound server to handle
voice calls. If yes, program the following parameters:
• Enable: Select this check box to enable Outbound Proxy. By default the flag is disabled.
• Server Address: Enter the Outbound Proxy Server's address. It may be an IP Address or Domain
name. A maximum of 48 Characters, including ASCII characters are allowed.
• Server Port: Enter the Outbound Proxy Server's Listening Port. The valid range for this is 1024-
65535. By default the Server Port is 5060.
• SIP Hardware Template: Assign a “SIP Hardware Template” to the SIP Trunk. The SIP Hardware
Template contains voice quality related features such as Voice Codec selection, Tx and Rx Gains,
Echo Cancellation, Jitter Buffer and, Fax-over-IP options and related parameters
First, check if the values in Template 01 fulfill the feature requirements of the SIP Trunks. Retain this
template, if it fulfills the feature requirements of all SIP Trunks and if the same features are to be
allowed to all SIP Trunks.
If different sets of SIP hardware features are to be allowed to different SIP Trunks, then prepare
separate SIP Hardware Templates and apply them on the SIP Trunks. To do this,
• Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator,
DID, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the behavior of
a Trunk. Apply a Trunk Feature Template to the SIP trunk. By default, Trunk Feature Template 01 is
applied on all SIP trunks as well as all other trunk types. Refer the topic “Trunk Feature Template” to
know more.
Click the 'Trunk Feature Template’ link to open the page. Check if the default Template 01 fulfills your
requirement for the SIP trunk.
If the default Template 01 does not fulfill your requirement, prepare another Trunk Feature Template94,
and enter the newly prepared Template number for the SIP trunk.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the SIP
trunk.
Cost Factor is used to grade the cost of routing calls from a SIP trunk, from 1 to 99; where 1 denotes
least cost and 99 denotes highest cost.
Assign a Cost Factor to the SIP trunk, for instance, 02 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '9' through the SIP trunk 01
only,
• You must first assign a Cost Factor (01-99) to SIP trunk 01, for example, 02.
• Click the 'Least Cost Routing - Number Based' link to open the page.
94. The default template is applied on the ports of all trunk types supported by ETERNITY. Changes to the default template will be
applied on all trunk types also. So, you are advised to prepare a new template and apply it to the desired trunk types.
All outgoing calls assigned Cost Factor trunk 02 will be made from SIP trunk 01.
• Simultaneous Calls: This parameter is for defining the number of Simultaneous Calls to be allowed on
the SIP trunk. ETERNITY ME and GE support up to 32 simultaneous calls, while ETERNITY PE
supports 16 simultaneous calls. You can configure as many simultaneous calls as supported on the
SIP trunk by the ITSP.
If the ITSP supports less than 8 simultaneous calls on SIP Trunks, you must program this parameter
accordingly. However, if you ITSP supports more than 8 simultaneous calls, you need not change this
parameter.
• If you have configured the above parameters for the desired VoIP Ethernet Ports, click 'Submit' to save
your configuration settings.
For such users, you may click the 'Advanced' button and configure the following parameters:
• Return Call to Original Caller (RCOC): Enable this flag if you want to apply the 'Return Call to Original
Caller' on this SIP trunk.
If this feature is enabled on the SIP trunk, the system will route calls returned by remote parties back to the
extensions that originally made the call from this Trunk (the original callers' extensions). To know more,
refer the feature description for “RCOC (Return Call to Original Caller)”.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the SIP trunk. There are
50 different templates to choose from. Each template can also be altered to suit your requirement and
preferences. Default: Template number 01 assigned to all SIP trunks.
• Station Advanced Feature Template: Assign a “Station Advanced Feature Template” to the SIP trunk.
There are 50 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. Default: Template number 01 assigned to all SIP trunks.
• Add 'rinstance' in Register: 'rinstance' is any random value which can be used by the VoIP Ethernet Port
to fetch its own contact binding, that is, to know the Registration Expiry Timer assigned by the server. By
default, this flag is enabled.
• Send REGISTER Message: With this parameter you can select whether or not the system should send
REGISTER message from the SIP trunk. By default, this flag is enabled allowing REGISTER message to
be sent from the SIP trunk.
• Allow OG Calls without Registration: This parameter is to be enabled to allow outgoing calls to be made
from the SIP trunk, even when the SIP trunk is not registered. If this flag is disabled the system will not
allow outgoing calls to be made if the status of the SIP trunk is 'not registered'. By default, this flag is
disabled.
• CLIR: Select this option if you do not want the CLI to be sent.
• SIP ID: You may select this option if you want the SIP ID programmed on the SIP Trunk to be sent as
CLI.
• Calling Party-wise: Select this option if you want to send the Calling Extension Number (the number
of the extension making the outgoing call through the SIP trunk) as CLI.
When reverse DDI is programmed on the SIP Trunk, the DDI number of the calling extension will be
sent, instead of its extension number.
If the calling extension has disabled the parameter ‘Send DDI as CLI’ in its Station Advanced Feature
Template, then its Pilot number configured in the Outgoing Reference Table will be sent as CLI.
If calling extension has enabled CLIR, no CLI will be sent on the SIP Trunk.
• Fixed Number: Select this option if you want a specific number to be sent as CLI. When you select this
option, you must also define the number to be sent as CLI.
You may select this option you wish to send any of your trunk line numbers as CLI on the SIP Trunk so
as to enable the called party to call back the calling party using this CLI.
Since it is not possible to call back a SIP ID, Fixed Number offers you a solution, using which you can
send a trunk line number as CLI on the SIP Trunk. Using this CLI, the called party can call back the
calling party.
The Fixed Number may consist of a maximum of 40 characters, including all ASCII characters.
By default, ‘SIP ID is set as the Send CLI option for all SIP Trunks.
When extension number of the calling extension is blank, and the ‘Send CLI Option’ programmed for the
the SIP Trunk is other than "SIP ID", then also SIP ID will be sent as CLI.
• Accept Anonymous Calls?: The flag is for accepting calls without CLI that land on the SIP trunk. By
default the flag is enabled. You may disable this flag to disallow calls without CLI on this SIP trunk port.
• Source Port IP Address: Select the Source Port IP Address for the SIP trunk. You may select from any of
the following options, as applicable to your installation scenario.
• Use VoIP Ethernet Port IP Address: Select this option, if the VoIP Ethernet Port on which the SIP
trunk is configured is connected directly to the public Internet.
• Use IP Address fetched using STUN: Select this option, if the VoIP Ethernet Port is located behind a
NAT router other than Symmetric. If you select this option, make sure to disable 'Outbound Proxy'.
• Use Router's Public IP Address: Select this option if the VoIP Ethernet Port is located behind a NAT
router (any type). If you select this option, make sure to disable 'Outbound Proxy'.
Select 'Yes' if the VoIP Ethernet Port is located on a public IP and you want outgoing calls to the SIP Client
located behind the NAT Router. OR if you need to receive incoming calls from the SIP Client located
behind the NAT router. By default, Use of Symmetric RTP is enabled.
• Digest Authentication: This flag is to be enabled, if you want the feature 'Digest Authentication' to be
enabled on the SIP trunk.
When Digest Authentication is enabled, incoming calls from callers will be allowed only after the callers
have authenticated themselves (with their User ID and Password). If the caller enters invalid authentication
information, the system will re-challenge authentication once more, and reject the call, if the authentication
attempt fails again.
When you enable Digest Authentication, make sure that the Digest Authentication Table is also configured
first. To do this,
• Scroll to go to 'Digest Authentication', and click this link to open the page.
• Enter the User ID to be authenticated along with its corresponding Password against each Index on the
Table.
• The User ID may be a maximum of 40 characters, including ASCII characters. The User Password may
be a maximum of 16 characters, including ASCII characters.
• If you have finished entering the User IDs and their corresponding User Passwords in this Table, click
'Submit' at the bottom of the page.
• Default Transport for Outgoing Message: The VoIP card of the ETERNITY supports three options for
transporting outgoing SIP messages:
• UDP: Outgoing messages are transported using UDP. By default this option is selected.
• TCP (Fallback to UDP): TCP is used for outgoing messages. However, if the TCP connection fails, the
system will attempt to send the message again over UDP.
These options are checked only if you have enabled SIP over TCP flag. If the SIP over TCP flag is
disabled, all outgoing SIP messages will be transported over UDP only.
• Call Budget: If you want to enable 'Call Budget on Trunk' feature on this SIP trunk, configure the following
parameters:
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 99999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 99999.
• Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
• Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset also using SA and SE
commands, referred to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
• Call Back: This parameter is related to the “Call Back on Trunk Port’ feature. If you want to enable the
'Call Back on Trunk Port' feature on this SIP trunk, configure the following parameters:
• Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
• Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds for all SIP Trunks.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote party
should be routed:
• DID
• PIN Authentication - Multiple Calls
• CLI Authentication - Multiple Calls
• CLI Authentication - Single Call - Answer Signaling
• Operator
• Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the ‘CLI number”. If you want the call back to be made to a different number, select ‘Alternate
Number’.
• Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
If you want a different set of number strings to be programmed for this SIP Trunk, select a different
Number List, and assign it to the SIP trunk port.
You may program the Incoming Number List either from the ‘Number List’ page or by clicking the
‘Incoming Number List’ link to reach the Number List page.
Refer the topic “Number List” to know more, and for configuration instructions.
• Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the ‘Incoming Number List’, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the ‘Outgoing Number List’.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all SIP trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this SIP Trunk port.
You may program the Outgoing Number List either from the ‘Number List’ page or by clicking the
‘Outgoing Number List’ link to reach the Number List page.
Refer the topic “Number List” to know more, and for configuration instructions.
• Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an Outgoing Trunk Bundle Group (OGTBG).
Select ‘Same port’ if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
• OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTBG that you define here for Call Back.
• Incoming Reference ID: Assign an Incoming Reference ID for the SIP trunk for Working Hours, Break
Hours, Non-Working Hours. By default, 00 is assigned as Incoming Reference ID for all three time zones.
• Outgoing Reference ID: Assign an Outgoing Reference ID for the SIP Trunk, from 00 to 99. By default,
00 is assigned as Outgoing Reference ID.
• Pause Timer (sec): This Timer is required for inserting delay while digits of a number string are out dialed
from the SIP trunk. The Pause Timer will be applicable when the letter 'P' is configured in the DTMF
number string which is to be out dialed as DTMF digits on the SIP trunk. The range of this timer is from 1 to
9 seconds. By default the Timer is set to 3 seconds.
• DTMF ON Time: This is the time for which the DTMF digit will remain ON, while being out dialed by the
ETERNITY. This parameter finds its application in the feature “Multi-Stage Dialing”. The range of this
timer is from 051 to 255 milliseconds. By default, ON Time is defined as 102 msec.
• Inter Digit Pause (msec): This is the time for which the ETERNITY will wait before dialing the
successive digits.
• Gateway Application - Answer Signaling: This parameter is to be programmed if the SIP trunk is being
used in a gateway application as a source port (from where calls originate). The calls originated on the
source port (SIP trunk) are routed using another Trunk port, the terminating port, which may be any trunk
port, like T1E1. When call made from the terminating port gets matured, this is signaled to the source port
in the form of DTMF digits.
• Use?: Enable this flag if you want the SIP trunk to be used in a Gateway Application.
• DTMF String: Program the DTMF digits to be sent to signal call maturity to the source port.
• If you have configured the above parameters for the desired SIP trunks, click 'Submit' at the bottom of the
page to save your configuration settings.
• For each SIP trunk (number), the following settings will be displayed:
• Status
• Registration Time
• Consumed Amount/Minutes/Calls
• Reset Consumed Budget (this is not a status indicator. It is for resetting the Consumed Call Budget
manually)
To configure Server Address of Outbound Proxy for the SIP trunk, dial:
• 7716-1-SIP-Outbound Proxy Server Address to configure Server Address for a single SIP trunk.
• 7716-2-SIP-SIP-Outbound Proxy Server Address to configure the same Server Address for a range
of SIP trunks.
• 7716-*-Outbound Proxy Server Address to configure the same Server Address for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Outbound Proxy Server Address is a string of a maximum 40 characters, with extended ASCII
character set.
• For Advanced Configuration of the SIP trunks, use the following commands:
To enable/disable 'Allow Outgoing Calls without Registration' flag on the SIP trunk, dial:
• 7745-1-SIP-Flag to enable/disable this flag on a single SIP trunk.
• 7745-2-SIP-SIP-Flag to enable/disable this flag on a range of SIP trunks.
• 7745-*-Flag to enable/disable this flag on all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Flag is
0 for restrict OG Calls without Registration
1 for allow OG Calls without Registration
Default: 1
The Index number should remain the same for each User ID and its corresponding password. For
example, if you configured a User ID at Index number 03, the corresponding password for this ID should
also be configured at Index number 03.
To configure Default Transport for Outgoing Messages for a SIP trunk, dial:
• 7731-1-SIP-Default Transport for OG Message to select the transport mode for a single SIP trunk.
• For SE Commands for configuring Call Budget on SIP Trunks, refer the feature description “Call
Budget on Trunk”.
To assign Call Back - Incoming Number List to a SIP Trunk port, dial:
• 7747-1 -SIP-Incoming Number List to assign a list to a single SIP trunk.
• 7747-2-SIP-SIP-Incoming Number List to assign the same list to a range of SIP trunks.
• 7747-*-Incoming Number List to assign the same list to all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Incoming Number List is from 01 to 16.
Default: 15
To assign Call Back - Outgoing Number List to a SIP Trunk port, dial:
• 7748-1-SIP-Outgoing Number List to assign a list to a single SIP trunk.
• 7748-2-SIP-SIP-Outgoing Number List to assign the same list to a range of SIP trunks.
• 7748-*-Outgoing Number List to assign the same list to all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Outgoing Number List is from 01 to 16.
Default: 16
To select Call Back From port for a SIP Trunk port, dial:
• 7749-1-SIP-Call Back From to select Call Back From for a single SIP trunk.
• 7749-2-SIP-SIP-Call Back From to select the same Call Back From for a range of SIP trunks.
• 7749-*-Call Back From to select the same Call Back From for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Call Back From is
1 for Same Port
2 for OGTB Group
Default: Same Port
To assign Incoming Reference ID to the SIP trunk for Working Hours, dial:
• 7722-1-SIP-IC Reference ID to assign reference ID to a single SIP trunk.
• 7722-2-SIP-SIP-IC Reference ID to assign the same reference ID to a range of SIP trunks.
• 7722-*-IC Reference ID to assign the same reference ID to all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
IC Reference ID is from 00 to 99. Default: 00.
To assign Incoming Reference ID to the SIP trunk for Break Hours, dial:
• 7723-1-SIP-IC Reference ID to assign reference ID to a single SIP trunk.
To assign Incoming Reference ID to the SIP trunk for Non-Working Hours, dial:
• 7724-1-SIP-IC Reference ID to assign reference ID to a single SIP trunk.
• 7724-2-SIP-SIP-IC Reference ID to assign the same reference ID to a range of SIP trunks.
• 7724-*-IC Reference ID to assign the same reference ID to all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
IC Reference ID is from 00 to 99. Default: 00.
To configure the DTMF Inter-Digit Pause Timer for the SIP trunk, dial:
• 7726-1-SIP-DTMF Inter-Digit Pause Timer to configure the timer for a single SIP trunk.
• 7726-2-SIP-SIP-DTMF Inter-Digit Pause Timer to configure the timer for a range of SIP trunks.
• 7726-*-DTMF Inter-Digit Pause Timer to configure the timer for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
DTMF Inter-Digit Pause Timer from 051 to 255 milliseconds. Default: 102 msec.
To configure DTMF String for Gateway Application-Answer Signaling on the SIP trunk, dial:
• 7728-1-SIP-Gateway Application DTMF String to configure string for a single SIP trunk.
• 7728-2-SIP-SIP-Gateway Application DTMF String to configure string for a range of SIP trunks.
• 7728-*-Gateway Application DTMF String to configure string for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Gateway Application DTMF String is a string of maximum 4 digits. Default: CCC.
• Exit SE mode.
• The Extended IP Phone for ETERNITY supplied by Matrix. The Matrix Extended IP Phone functions just
like a DKP.
• Any standard SIP phone or SIP enabled device, such as an IP phone, a Soft phone, an Analog phone
adapter.
SIP Extensions function like any normal DKP/SLT extension of the ETERNITY. SIP Extension users can make and
receive calls to any extension user of the ETERNITY as well as any external numbers over PSTN, GSM, VoIP and
E&M lines, depending on the “Logical Partition” configured in the System.
The number of SIP Extensions supported by the different models of ETERNITY are:
• ETERNITY ME: 999 SIP Extensions
• ETERNITY GE: 500 SIP Extensions
• ETERNITY PE: 50 SIP Extensions
SIP Extensions are a licensed feature. Decide the number of SIP Extensions you will require and buy the
license. Refer the topic “License Management” to know more.
You can register a SIP Extension at three different locations as a single SIP Extension for Call Forking.
• VoIP Port No.: This is the software port number of the VoIP Card.
• Hardware Slot: This is the number of the hardware slot in which the VoIP Card is located.
• Name: This is the name you have assigned to the VoIP Card, when you configured the VoIP Port
parameters.
• As the Source Port IP Address, select the NAT Traversal mechanism for SIP messages from the
following options:
• Use VoIP Ethernet Port IP Address: Select this option if your VoIP Card is not located behind a
NAT Router.
• Use Router's Public IP Address: Select this option if your VoIP Card is located behind a NAT
Router, and you have set 'Router's Public IP Address' as the NAT Traversal mechanism in the “VoIP
Port Parameters”.
• Use IP Address fetched using STUN: Select this parameter if your VoIP Card is located behind a
NAT Router, and you have set 'Use IP Address fetched using STUN' as the NAT Traversal
mechanism in the “VoIP Port Parameters”.
• You may set the Minimum Registration Timer (sec), as required. This is the Minimum Expiry Timer,
which the User Agent should send in its REGISTER request. If the expiry value in the REGISTER
message is less than this value, the request will be rejected. The valid range of this timer is from 10 to
99999 seconds. By default, it is set to 45 seconds.
• In the Private Key field, enter the MD5 authentication key the VoIP Card of ETERNITY should use to
encrypt/decrypt the SIP messages. The Private Key may consist of a maximum of 24 characters. By
default, the field is blank.
• You can restore the default values of any one or all the parameters on this page by clicking the Default
One and the Default button respectively.
The parameters of the SIP Extension number you selected will appear on this page.
• In the VoIP Port No. field, select the software port number of the VoIP Port to which you want to assign the
SIP Extension. For example, you want to assign SIP Extension 1 to VoIP Port number 2, select 02 as port
number from the list.
• Select the Use SIP Extension check box to enable the SIP extension. Default: enabled.
You may clear this check box, when you want to deactivate the SIP extension.
• In the Name field, enter a name for the SIP Extension, which may be the name of the person who will use
the SIP Extension or the name of a Department. The name you enter here will be displayed as the Caller
ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
If no name is assigned to the SIP Extension, the system will display the name received in the INVITE from
the SIP Extension user for making outgoing calls.
• Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP Extension with the
Registrar of the VoIP Card. It is the number with which you can call the SIP Extension. Any extension user
of the ETERNITY can call a SIP Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of
each SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits
from 0 to 9 and the characters * and # are allowed.
By default, SIP Extensions are assigned the following SIP ID (Access Codes):
001 3301
002 3302
: :
500 4300
• In the Authentication ID field, enter the number which you want the VoIP Card's Registrar Server to use
for user authentication of the SIP messages received from the SIP Extension. The number may be a string
of maximum 6 digits. Any combination of digits from 0 to 9 and the characters * and # are allowed in the
string.
Authentication ID is applicable only if any of the 'Authentication' of SIP Message Options, namely
REGISTER or INVITE or SUBSCRIBE or PUBLISH, is enabled. Else, Authentication ID will not be used.
• In the Authentication Password field, enter the password to be used by the VoIP Card to authenticate
the SIP messages received from the SIP Extension. You can enter a maximum of 24 digits as password.
The valid digits for the password are 0 to 9, * and #. The default Authentication Password is 1234.
• In the Call Appearances field, define the maximum number of simultaneous calls that the SIP Extension
user should be allowed to make/receive. You can set up to 10 call appearances for a SIP Extension. By
default, Call Appearance is set to 1.
When Call Appearance is set to 1, the SIP Extension can make/receive only 1 call at a time.
• Under Authentication, enable Authentication of any or all of the following SIP Message Options by
selecting the respective check boxes:
• REGISTER Request
• INVITE Request
• SUBSCRIBE Request
By default, the SIP Message Options REGISTER, INVITE, SUBSCRIBE are enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed, when any of the above
SIP Message Options are enabled.
• To provide voice mail facility to the SIP Extension, select the Enable Voice Mail Subscription check box.
Default:
• To allow the SIP Extension to monitor the status of another extension, device or number, select the Enable
Busy Lamp Field95 Subscription check box. Default:
• To allow the SIP Extension to show his/her Presence Status to other SIP Extensions, select the Enable
PUBLISH check box. Default: disabled.
If you enable PUBLISH subscription for a SIP Extension, you must also enable Authentication. By
default, authentication is enabled.
The SIP Extension for which you have enabled Presence Subscription will be able to view Presence of
only those SIP Extensions which have PUBLISH enabled.
• If you want to register this SIP Extension from more than one location, enable the Allow Multiple
Registration check box. Default:
You can register a SIP Extension at three different locations as a single SIP Extension for Call Forking.
• If the SIP Extension is a Matrix Extended IP Phone, select a Key Template for the extension. The Key
Template may be of the Operator, Executive, Guest or Hotel, according to the key map you want to
assign to this extension. Default: Operator’s Template.
Like the DKP, the Extended IP Phone will function as Operator, Executive, Hotel Attendant, and Hotel
Guest extension, according to the key map template you assign. For example, if the Extended IP Phone is
to be used by the Operator, select Operator's Template. The phone will be assigned the key template with
the special features required by Operators, such as more DSS keys for Trunk Access and Call
Appearances, a Call Release Key, etc.
Similarly, if the user of the Extended IP Phone is a Hotel Attendant, select 'Hotel Attendant's Template'.
The key map with the specific Front Desk User features such as Check-In, Check-Out, Guest In/Out,
Change Room Clean Status, Room Shift, will be automatically assigned to the Extended IP Phone.
To know more about key templates, and for instructions on customizing them, read the topic “DSS Keys
Programming”.
If you want to customize the key map of this Extended IP Phone instead of applying a key template, select
the option Personalized, and configure the Phone Key Settings. See “Matrix Extended IP Phone Settings”
for instructions.
The template you assign will be applied on the Extended IP Phones registered at all three locations.
• Assign a SIP Hardware Template to the SIP Extension. Default: 01. The “SIP Hardware Template”
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Extensions as well as to
SIP Trunks.
95. Busy Lamp Field (BLF), a typical feature supported by PCM/TDM PBX and Key Telephone Systems, is also supported on SIP
Extensions.
In PCM/TDM PBX and Key Telephone Systems, this feature is typically used by the Operator to monitor the status of another
extension, that is, whether it is available, ringing or busy. The status of the other extensions is indicated on the special function
keys programmed on the Operator's console. This helps the Operator decide whether to place the call, or transfer the call to that
extension, or pick up the call ringing on that extension.
With BLF Subscription enabled on the SIP Extension, the user can monitor the status of another extension, device or number
within the same Proxy Domain.
If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
• Assign a Station Basic Feature Template to the SIP Extension. Default: The “Station Basic Feature
Template” has a set of features like Time Table, Class of Service, Toll Control, Operator, Storage of
Incoming and Outgoing Calls, Outgoing Trunk Bundle groups. There are 50 different templates to
choose from. Each template can also be altered to suit your requirement and preferences.
If the default Station Basic Feature Template 01 fulfills the feature requirements of the SIP Extension
(“Class of Service (COS)”, “Toll Control”, “OG Trunk Bundle Group”, etc.) retain this template, you may
also customize this template. If you want to assign a different set of features to this SIP Extension,
prepare a different Station Basic Feature Template and apply it to this extension. To do this,
• Select the number of the Template you customized, Template 05, in the Station Basic Feature
Template field.
Also, see the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on extensions.
• Assign a Station Advanced Feature Template to the SIP Extension. Default: Template 01. The
“Station Advanced Feature Template” has a set of advanced features for extensions such as Message
Wait Notification and Alarm Notification settings, Routing of Incoming DID Calls, Call Duration Control,
Floor Service, etc. There are 50 different templates to choose from. Each template can also be altered
to suit your requirement and preferences.
Check if the default template fulfills the feature requirements of the SIP Extension by clicking the
Station Advanced Feature Template link.
You may retain this template and customize it further, or customize another template if a different set of
features are to be allowed to this SIP Extension. To customize/prepare another template,
• Select the Template number, for example 02, and customize this template.
• In the Station Advanced Feature Template field, select the number of the template you
customized.
Also see the topic “Station Advanced Feature Template” for instructions on customizing these
templates and applying them on the extensions.
Call Pick Up allows the SIP Extension user to 'pick up' (answer) calls ringing on any other station, by
dialing a feature code, without physically going to the ringing station. It also allows incoming calls for the
SIP Extension to be answered by the other extensions assigned the same Call Pick-Up group.
For this to work, both the ringing station and the station picking up the call must be in the same 'Call Pick
Up Group'. Refer “Call Pick Up” for instructions on how to create groups. You can create as many as 99
groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SIP Extension in this field.
Advanced Settings
• If you want to provide other features like Personal Directory, Priority, or assign a Station Type to the SIP
Extension, click the Advanced button at the bottom of the page.
• If using the system in the Hotel Mode, select the Station Type for the SIP Extension as Administration or
Guest.
• You may assign a Personal Directory number to the SIP Extension. Default: 00.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes ("Out Going Trunk Bundle Group Index"). The Personal
Directory is necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
When a Personal Directory is assigned to a SIP Extension, make sure you also configure this directory.
The Personal Directory can be programmed by the SIP Extension users and by the System Engineer.
Refer the topic “Abbreviated Dialing” for instructions on programming the Personal Directory.
Each station of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being lowest
Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description “Priority”.
If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
You can register three Matrix Extended IP Phones at three different locations as a single SIP Extension.
If you have connected more than one Matrix IP Phone as a SIP Extension, configure their settings as Location 1,
Location 2 and Location 3.
• Select the check box Enable Matrix IP Phone Mode. Default: Disabled.
• Enter the MAC Address96 of the Matrix Extended IP Phone connected at this location in hexadecimal
format: 00:50:C2:55:B0:10. Default: blank.
ETERNITY validates the IP-Phone on the basis of the MAC Address, and provides configuration on
validation.
96. MAC address is the address of the electronic hardware devices such as a computer, which is hard-coded into the device during
manufacture and cannot be modified. No two devices can have similar MAC address and thus it uniquely identifies your phone.
MAC address is assigned as per the IANA standard. The MAC Address of the phone will be used as source MAC address on all
Ethernet frames.
• Select the appropriate Registrar Server IP Address to register the IP phone with the SIP Registrar of
ETERNITY, according to your installation scenario:
• If the Extended IP phone is in the same WAN network as ETERNITY, select Use WAN Port IP
Address as Registrar Server IP Address.
• If the Extended IP phone is in the same LAN network as ETERNITY, select Use LAN Port IP Address
as Registrar Server IP Address.
• If the Extended IP Phone is connected in the Global Network and ETERNITY is located behind a
Router, select Use Router/STUN's IP Address as Registrar Server IP Address.
Make sure the Router’s Public IP Address is configured in the Network Parameters.
• If the Extended IP Phone is connected in the Global Network and ETERNITY is located behind a NAT
Router, and STUN is programmed, select Use Router/STUN's IP Address as Registrar Server IP
Address.
• If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
Registrar Server IP Address.
• To set the call progress tone generation standards of the country where the Matrix IP Phone is installed,
select the Call Progress Tone - Region. Default: Region 1.
• To display the Date and Time of the country where the Matrix IP Phone is installed, select the Date and
Time - Region. Default: India.
• If you want to enable Daylight Saving Time (DST) on the phone, set Apply DST? to Yes. Default: No.
The Daylight Saving Time convention followed in the country/region you selected will be automatically
applied. The IP phone will change its date and time settings according to the DST convention of the
selected country/region.
• Select the CO CLIP Pattern for the Extended IP phone. This is the type of Calling Line Presentation on the
phone for incoming calls from trunks. You can select any of these options:
• Name Only (only the name of the caller will be displayed).
• Number Only (only the number of the caller will be displayed).
• Number + Name (both the name and the number of the caller will be displayed).
ETERNITY provides language support for English, French, German, Spanish, Portuguese, and Italian on
the Matrix IP Phone. When you select any of these languages, all the prompts and command strings will
appear in the selected language.
• If you selected Ring after a delay as Ringer Mode, set the Ring Delay Timer (sec), if required, to the
desired value.
The Ring Delay Timer is the time in seconds the system waits on receiving a call before ringing on the IP
phone. The range of this timer is 0 to 99 seconds. Default: 10 seconds.
• If you want to enable Ringer Auto Acknowledge mode, set the Acknowledge Timer (sec) to the desired
value.
The Ringer Auto Acknowledge mode determines when to stop the ring on the IP phone. There are two
options for Ringer Auto Acknowledge:
• Stop only when the call is answered.
• Stop after a delay.
To stop the ring on the IP phone after a delay, the Acknowledge Timer must be configured. The range of
this timer is 01 to 99 seconds. Default: 00 seconds.
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
• To assign the Ring Destination for the IP phone, select the desired destination for Play Ring on. You may
choose
• Speakerphone: The ring will be played on the Speakerphone.
• Headset: The ring will be played on the Headset.
Default: Speakerphone.
When you select the Headset as the destination, make sure that you set the flag ‘Headset Connected?’ to
Yes, connect a Headset to the IP Phone.
• Select the desired Ring Tune according to your/IP phone user’s preference. Default: 1.
• Set the Ringer Volume to the desired level, from 0 to 7, according to your preference. Default: 5.
• To increase/decrease the volume of outgoing speech (Transmit Gain) on the handset of the IP phone, set
the Handset Transmit Volume to the desired level, from 0 to 7. Default: 4.
• To increase/decrease the volume of incoming speech (Receive Gain) on the handset of the IP phone, set
the Handset Receive Volume to the desired level, from 0 to 7. Default: 4.
• To increase/decrease the volume of outgoing speech (Transmit Gain) on the headset of the IP phone, set
the Headset Transmit Volume to the desired level, from 0 to 7. Default: 4.
• To increase/decrease the volume of outgoing speech (Receive Gain) on the headset of the IP phone, set
the Headset Receive Volume to the desired level, from 0 to 7. Default: 4.
• To change the Receive Gain of the Speakerphone MIC Volume, set Speaker Receive Volume to the
desired level, from 0 to 7. Default: 4.
• To use a Headset with the IP phone, set Headset Connected? to Yes. Default: No.
Make sure that you connect a Headset to the Extended IP phone, if you select Yes.
• Select the Auto Answer check box to enable this feature on the Extended IP phone. Default: Disabled.
When you set the “Auto Answer” feature on the Extended IP phone, the phone goes OFF-Hook
automatically after a preset period of time, without the extension user having to pick up the handset or
press the speaker or headset key. When you enable Auto Answer, you must configure the Auto Answer
Timer.
• If you enabled Auto Answer on the phone, set the Auto Answer Timer (sec) to the desired value.
This timer defines the time in seconds that the IP phone should wait before going OFF-Hook to auto
answer a call. The range of this timer is 1 to 9 seconds. Default: 1 second.
• Adjust the Backlight brightness of the phone’s LCD display, by setting the LCD Backlight Level to the
desired value, from 1 to 4. Default: 3.
• Set the Back Light Off Timer (sec) to the desired value, if required, from 000 to 999 seconds. Default: 10
seconds.
• Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you. Default: 3.
• To personalize the key map of the Extended IP Phone97, click the Phone Key Settings link.
97. To personalize the phone key settings, you select Personalized Key Template for the SIP Extension.
The Operator function is a Feature, so select the option FEATURE from the Select Function Type list
box.
From the Select Offset drop down list, all the features that can be assigned to keys are listed.
• Select Operator from the list of features in the Select Offset box.
• Click OK.
• To take a second example, if you want to assign Remote DND to the key currently assigned TWT 2 key,
click the key.
• In the Select Offset box, select the option Set DND for remote station.
• Click OK. The box closes. Remote DND feature will appear in abbreviated form as R-DND on the key
label.
• Follow the same instructions to assign features to other DSS keys. Selecting the appropriate Function
Type and the Offset for each feature/function.
If you want assign a feature, select FEATURE as function type, and select the desired feature as Offset.
If you want to use the key to call a DKP or a SIP extension, select DKP or SIP Extension as Function
Type and select the number of the extension as Offset.
To assign direct access to a mobile trunk, select MOBILE as Function Type and the desired port number 1
or 2 as Offset.
To assign direct access to a SIP Trunk, select SIP as Function Type and the desired trunk number from 1
to 4 as Offset.
You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
If you have upgraded your SETU VP248 to an Extended IP Phone with firmware V5Rx, the capsense key
labels listed in the table below will have the following functions:
DND Forward
Reject Release
SIP/RTP Ports
• SIP Listening Port: This is the port on which the IP phone listens for SIP messages over TCP. This
port is also used as the source port for sending SIP messages to the remote peer. The valid range for
this port is 1024-65534. Default: 5060.
• RTP Listening Port: This is the port on which the IP phone listens for SIP messages over TCP. This
port is also used as the source port for sending RTP packets. This port is also used as the source port
for sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default: 8000.
Quality of Service
• Set the SIP Quality of Service (QoS) for SIP signaling as:
• If the Extended IP phone is connected behind a NAT router, configure NAT Keep Alive.
• Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
• Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
The time period you define should be less than the binding timer of the router.
• SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user
disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds that the phone waits for final response after
receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the IP phone terminates the call
process and gives error tone to the user. The range of SIP Provisional Timer is 10-180 seconds.
Default: 60 seconds.
• General Request Timer (sec): This is the time in seconds for which the phone waits for response of a
transaction request. This timer starts on initiating a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the phone clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.
Debug
• To debug using Syslog Client supported by the Extended IP Phone, configure Debug parameters:
When the Debug flag is enabled, the phone will send the debug messages to the Syslog Server IP
address. Debug report can be viewed on the Syslog Server or any other application which can capture
the Syslog messages/debug sent by the phone.
• Enter the IP Address and port of the remote Syslog Server and as Syslog Server Address: Port.
The address of the Listening Port of the Syslog Server is from 1024-65535. Default: 514. Syslog uses
the UDP as transport protocol and listens on the port 514 (the default listening port).
• You may select the Debug Level from the following options, by selecting the respective check box:
• SIP
• System
• Hardware
• Call
• Menu
• User Interface
You may select any or all of these debug levels. The Syslog Client will send only the debug messages
for the selected level to the remote server on the IP network. For example, if the debug log of 'Call's is
required, you can select this option, and disable all others.
• If you have completed the configuration of the Matrix IP Phone Settings at Location 1, follow the same
steps as described above to configure the Matrix Extended IP Phone at Location 2 and Location 3.
General Parameters
To select Source Port IP Address, dial:
• 7830-1-VoIP Port-Source Port IP Address Option
Where,
VoIP Port is the software port number of the VoIP Port from 01 to 16.
Source Port IP Address Options are
1 for Use IP Address of VoIP Ethernet Port.
2 for Use IP Address Fetched using STUN.
3 for Use Router’s Public IP Address.
By default, Use IP Address of VoIP Ethernet Port is selected.
Channel number for all function types other than BRI and PRI is 00
For the complete list Function Type and Function Number see “DSS Keys Programming”. The Key
Numbers of the MATRIX Extended IP Phone is the same as EON48 described under “Customizing Key
Templates using a Telephone”.
• Exit SE mode.
• The SIP Extension Status page will open and display the following for each SIP Extension
• Contact 1
• Contact 2
• Contact 3
The ETERNITY ME and GE support E&M interface. This interface is not supported on ETERNITY PE.
If you have correctly installed the E&M Cards and observed the Reset Cycle, you may now program the E&M Ports
“Using Jeeves” or a Telephone, depending on your installation scenario.
OR
• a Trunk - works like a trunk interface when any of the stations of the PBX makes an outgoing call through
it.
OR
• a Tie Line - takes on a dual personality: functioning as both a station and a trunk. The E&M port works like
a station interface for incoming calls. It works like a trunk interface when any station makes an outgoing
call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
“PLCC-An Introduction” to know more.
• E&M Trunk Seizure Type98: Immediate, Immediate with Ack, Immediate + Wink, Immediate with
Ack+Wink (MFCR2)99, Seizure Pulse, Seizure Pulse + Wink, Express, and Radio.
• Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The E&M Interface (Type IV and Type V connection) and the Speech Interface (2-wire speech or 4-wire)
are selected at the time of installation by changing the Jumper settings.
98. This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
99. Currently supported only on ETERNITY GE E&M4 Card.
Configure the following parameters for each E&M port on this page:
• E&M No.: This non-editable field is the software port number of the E&M Port. Refer the topic
“Software Port and Hardware ID” to know more.
• Hardware Slot - Port: 'Slot' is the number of the number of the Universal Slot in which the E&M Card
is inserted. 'Port' is the number of the E&M hardware port on which the Tie Line equipment (PBX,
Router, Leased Line, etc.) is connected.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the E&M software ports.
For example: if you have inserted the card E&M8 in Slot 05 and E&M4 card in Slot 06 of ETERNITY
ME16S, the system will assign the Hardware Slot 05 and port numbers 01-08 to the E&M Software
Ports from 001 to 008 respectively. The system will assign hardware Slot 6 and port numbers 01-04 to
the E&M Software Ports 009 to 012. Refer the topic “Software Port and Hardware ID” to know more.
However, if required, you may change the Hardware Slot and Port assigned to the E&M software port.
In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
• Enable Port: This flag is for enabling or disabling an E&M port. When an E&M port is disabled, neither
incoming nor outgoing calls can be made from that port.
• Name: You may assign a Name to the E&M Port. Whenever there is an incoming call on this Port, this
name will be displayed on the destination station, while receiving the call.
• E&M Feature Template: This parameter is applicable to all E&M Ports. The E&M Feature Template is
a complete set of E&M features to be applied on E&M Ports according to their 'Orientation Type',
whether they are Stations, Trunks or Tie-Lines.
By default E&M Feature Template 01 is applied on all E&M Ports. This template has 'Station' as the
default Orientation Type.
If all the E&M Ports are to be programmed as 'Stations' retain this template.
If all the E&M Ports are to be programmed as 'Trunks' use the default E&M Feature Templates 09 and
10 which have 'Trunks' as Orientation Type.
If some of the E&M Ports are to be programmed as Stations, some as Trunks and yet others as Tie
Lines, prepare different E&M Feature Template for each Orientation Type and apply them to the related
ports.
• Apply the E&M Feature Template you customized to the E&M Port by entering the template number
in the 'E&M Feature Template' field of this port.
• Repeat the same steps to customize another template and apply it to another E&M Port.
Refer the topic “E&M Feature Template” for more details on customizing the templates and applying
them on E&M Ports.
• Trunk Feature Template: This parameter is relevant only if the E&M Port is to be programmed to
function as a Trunk or a Tie-Line100. To know more, refer “E&M Feature Template”.
Assign a “Trunk Feature Template” to the E&M Port. A Trunk Feature Template is a set of features like
Time Table, Operator, DID, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that
defines the behavior of a Trunk. Apply a Trunk Feature Template to the TWT Trunk port.
100. To program the E&M Port as a Trunk or a Tie-Line, you must set the 'Orientation Type' of the E&M Port to 'Trunk' or 'Tie-Line' in the
“E&M Feature Template” applied on the port.
Click the link 'Trunk Feature Template to open the page. Check if the default Template 01 fulfills your
requirement for the E&M Trunk port.
If not, you may prepare a different Trunk Feature Template and apply on all E&M Ports. For this,
If the default Template 01 does not fulfill your requirement,
You may prepare a different Trunk Feature Template and apply on all TWT Ports. For this,
• Go to the E&M Software Port Number you want to which you want to assign the Template you
prepared.
• Enter the number of the Template you prepared (04) in the 'Trunk Feature Template' field.
You may also prepare different Templates for different E&M Ports, for example Template 04 for certain
ports, Template 05 for others. In which case, for each E&M Port, enter the number of the template you
have prepared for that port.
To know more about customizing templates, refer the topic “Trunk Feature Template”.
• Station Basic Feature Template: This parameter is applicable on when the E&M Port is to be
programmed to function as a Station (Orientation type = Station or Tie-Line).
Assign a “Station Basic Feature Template” to the E&M Port functioning as a Station. By default, Station
Basic Feature Template 01 is assigned to all stations, that includes SLT and DKP ports.
Check if the default template fulfills the feature requirements (like “Class of Service (COS)”, “Toll
Control”, “OG Trunk Bundle Group”, etc.) of the E&M Ports functioning as Stations.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all E&M (station) ports, retain Template 01.
If different sets of features are to allowed to different E&M (station) Ports, then prepare separate
Station Basic Feature Templates and apply them on the ports. To do this,
• Click the link 'Station Basic Feature Template' to open the page.
• Enter the number of the Template you customized, Template 05 in the 'Station Basic Feature
Template' field of the E&M Port (for example: E&M No. 003) on which you want to apply this
template. If you want to apply this template to other ports too, like E&M No. 004, 005, and 006,
assign the Template 05 to all these ports.
• Repeat the same steps to customize and assign a different Template to another E&M port.
Also, refer the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on the ports.
• Station Advanced Feature Template: This parameter is applicable only when the E&M Port is to be
programmed to function as a Station.
By default Station Advanced Feature Template 01 is assigned to all stations, that includes SLT and
DKP ports as well as E&M ports with the orientation type 'Station'.
Check if this default template fulfills the feature requirements of the E&M Ports (with 'station' as
orientation type) by selecting the 'Station Advanced Feature Template' link.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all E&M (station) ports, retain Template 01.
If different sets of features are to be allowed to different E&M (station) Ports, then prepare separate
Station Advanced Feature Templates and apply them on the ports.
To do this,
• Click the 'Station Advanced Feature Template' link to open the page.
• Enter the number of the Template you customized, Template 03 in the 'Station Basic Feature
Template' field of the E&M Port (for example, E&M No. 003) on which you want to apply this
template. If you want to apply this template to other ports too, like E&M No. 004, 005, and 006,
assign the Template 03 to all these ports.
• Repeat the same steps to customize and assign a different Template to another E&M (Station) port.
• Priority: This parameter is applicable only when the E&M Port is to be programmed to function as a
Station101. To know more, refer “E&M Feature Template”.
Each station of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever a station (phone) with higher priority calls a
station with lower priority, a triple ring is placed on the called station. To know more, read the feature
description “Priority”.
By default, the Priority of all E&M Ports functioning as Stations is set to '5'. So, decide what Priority
Level you will assign to each of the E&M Ports functioning as Stations and set the desired level for
each port.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the E&M
Trunk port.
Cost Factor is used to grade the cost of routing calls from an E&M trunk (orientation), from 1 to 99;
where 1 denotes least cost and 99 denotes highest cost.
Assign a Cost Factor to the E&M Trunk port, for example, 03 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the E&M Trunk Port
001 only,
• You must first assign a Cost Factor (01-99) to E&M Trunk Port 001, for example 03.
• Click the 'Least Cost Routing - Number Based' link to open the page.
• Enter '6' in the 'Number' column, Cost Factor '03' as Preference 1, 2, 3 and 4.
All outgoing calls assigned Cost Factor trunk 03 will be made from E&M Trunk Port 001.
• If you have completed configuration of all the above listed E&M Parameters, click 'Submit' at the
bottom of the page to save your changes.
101. Recall that E&M ports can function as trunks, as stations and have both functions. When an E&M Port is programmed as a Station
Interface, it can only receive incoming calls. To program the E&M Port as a Station, you must set the 'Orientation Type' of the E&M
Port to 'Station' in the E&M Feature Template applied on the port.
This command is applicable only when the E&M Port is configured to function as a station!
• Exit SE mode.
A magneto telephone is a local battery telephone set, in which signaling current is provided by a hand generator,
usually a magneto. The hand generator, commonly referred to as 'crank', is located on the right hand side of the
telephone set and is turned to produce energy to ring other phones.
Tip and Ring wires of the magneto phone are dry, that is, they do not carry battery voltage when idle or in speech.
This means the wires carry only AC voice signals and not the DC voltage of the battery.
The Battery is used to power the microphone only and ringing generator is used to provide ring voltage to alert the
other phone.
Magneto Telephones are widely used by defense establishments as field phones in front lines, and by other
establishments such as railroad companies (signaling emergencies, crossings, etc.), electric utilities, pipeline
companies, who need to have their networks at places that are too remote to be serviced by public telephone
networks.
The Magneto Interface is supported on ETERNITY ME 10S and ME16S. A maximum of 128 Magneto Ports are
supported. The availability of ports depends on the variant of ETERNITY ME and the number of Magneto Cards
installed in the system.
• Ring is played on the Magneto Field Telephone connected to the Magneto Port.
• The extension user gets Ring Back Tone for the duration of the Ring Back Tone Timer.
• The extension user must press # or the Magneto Ring Enable (MRE) Key (on the DKP) before the expiry of
the Ring Back Tone Timer (programmable; default: 45 seconds) to check if the Magneto User has
answered the call102.
• The system considers pressing of the # or MRE Key as call maturity and connects the speech path
between the Magneto Field Telephone (connected to the Magneto Port) and the extension phone.
• If the Magneto User has answered the call, the extension user may start speech on hearing the Magneto
User's voice.
• If there is a period of silence, the extension user may press the # or MRE Key again to generate Ring on
the Magneto Telephone, and press the # and MRE Key once again during the Ring Back Tone Timer to
check speech with the Magneto User.
102. As there is no Answer Signaling or Call Disconnect feature on the Magneto line, there is no way for the Extension user to know
whether speech has been established with the Magneto User, but to wait to hear the Magneto User's voice.
• The extension user either goes ON-Hook while in speech. Error Tone will be played to the Magneto
user for the duration of the Error Tone Timer and the system releases the Magneto Port.
• The Magneto User cranks the hand generator (Ring Down Signal) while in speech to disconnect the
call. The system detects the Ring Current and releases the Magneto Port.
• If the Magneto port is busy, the extension user will hear the Busy Tone.
• If the extension user goes ON-Hook while the Magneto Phone is ringing, the system will stop the ring
on the Magneto phone.
• The system detects the ringing current. If the ring current is present for more than 500msec, the system
treats it as a Call request from the Magneto Port.
• The system rings the Operator extension assigned to the Magneto Port for the duration of the Ring Back
Tone Timer and plays Ring Back Tone to the Magneto Telephone user.
• If the Operator goes OFF-Hook, before the expiry of the Ring Back Tone Timer, speech is established
between the Operator and the Magneto User.
• The Operator can now transfer the call to another extension or external number.
• The Operator can go ON-Hook while in speech to disconnect. Error Tone will be played to the Magneto
user.
OR
• The Magneto User can cranks the hand generator again (Ring Down Signal) while in speech to
disconnect the call.
• The system releases the Magneto Port and the port with which the Magneto User is in speech.
• If neither the Operator nor any of the two Magneto station users in speech disconnects the call, the system
will automatically disconnect the call if silence (no speech) is detected for more than a specified duration of
time.
• For this, the flag 'Enable Silence Detection on Magneto' must be enabled in the System Parameters and
the 'Magneto Silence Disconnection Timer (default: 60 seconds) must be programmed. When Silence
Disconnection is enabled on Magneto port, the system will start the Magneto Silence Disconnection Timer
as soon as it detects silence. If the continuous silence is detected till the expiry of the timer the system
considers the conversation as over and releases the Magneto stations. The call is disconnected. However,
if speech is detected during this timer, the timer will be stopped.
• The call can be disconnected also using Forced Release by any extension user (Operator or any other
extension user) having the feature 'Forced Release' in its Class of Service.
• The Magneto user may crank the hand generator again to send Ring Current on the expiry of the Ring
Back Tone Timer. However, if the Magneto user sends the Ring Current again during the Ring Back
Tone Timer, the system will treat it as a call disconnect event. It will stop ringing the Operator's
extension and release the Magneto Port.
• The Magneto Ring Enable (MRE) Key should be programmed on the DKP extensions that are to be
allowed access to call Magneto ports.
• On SLT extensions and ISDN Terminals, the # key will serve the function of the MRE key.
• Magneto Port No.: This non-editable field is the number of the software port of the Magneto port.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the Magneto software ports.
However, if required, you may change the Hardware Slot and Port assigned to the Magneto software
port. In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
• Enable Port: This flag is for enabling or disabling a Magneto port. When a Magneto port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by selecting the check
box.
• Access Code: Assign Station Access Codes to the Magneto Port. Station Access Codes are
commonly referred to as Extension Numbers. These may be number strings of a maximum 6 digits,
which are to be dialed to call the Magneto port to which they are assigned.
All Magneto ports are assigned the following Station Access Codes as default.
001 2701
002 2702
003 2703
: :
: :
128 2828
You may either apply the default Station Access Codes to the Magneto ports or assign them according
to your requirement and preferences.
A maximum of six characters are allowed in an Access Code. These may be any combination of digits
between 0-9, and the characters *, #, A, B, C, D, F, P and 'Blank'.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash
with any other Access Code in the 'Dial' phase. Refer the topics “Access Codes” and “Conflict
Dialing” to know more.
• Name: Assign a 'Name' to the Magneto port. The name may be of the person who will use the Magneto
telephone or the name of the department or location of the telephone. This name will be displayed on
the LCD of the Operator/extension user's phone, if it is equipped with Caller ID.
• Station Basic Feature Template: As the Magneto Port functions as a station, assign a “Station Basic
Feature Template” to the Magneto port.
Only the following features of the Station Basic Feature Template are applied on the Magneto Port:
• Time Table
• Operator
• Class of Service
By default, Station Basic Feature Template number 01 is assigned to all station types of the ETERNITY
(SLT, DKP, ISDN Terminals, E&M Lines with Station as Orientation Type). Template 01 is also applied
on Magneto ports by default.
Check if the default settings of the features applied on the Magneto ports (Time Table, Operator, Class
of Service, Storage of Outgoing and Incoming Calls) match your requirements for the Magneto ports. If
yes, retain the default Station Basic Feature Template 01.
If you want to change any of the feature settings in for the Magneto Ports, you may prepare a different
Template103, for example, Template 14 and apply it on the Magneto Ports.
Also, if different feature settings are to be applied on different Magneto Ports prepare separate Station
Basic Feature Templates and apply them on the ports. To do this,
• Click the 'Station Basic Feature Template' link to open the page.
• Customize the Magneto Port related features (listed above) in Template number 14 and click
'Submit' to save.
• Enter the number of the Template you customized, Template 14 in the 'Station Basic Feature
Template' field of the Magneto Port, for example: MAG-001, on which you want to apply this
template. If you want to apply this template to other ports too, like MAG-002, 003, and 004, assign
the Template 14 to all these ports.
103. This is recommended because changing the values of the default Template will be applied on all other station types to which the
Template is assigned.
Refer the topic “Station Basic Feature Template” to know more about customizing the templates and
applying on the ports.
• Station Advance Feature Template: Assign a Station Advanced Feature Template to the Magneto
Port.
Only the following features in Station Advance Feature Templates are applied on Magneto Ports.
By default Station Advanced Feature Template 01 is assigned to all stations of the ETERNITY, which
also includes DKP ports, SLT ports, ISDN Terminals and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the Magneto Ports by opening the link
'Station Advanced Feature Template'.
Check if the default settings of the features applied on the Magneto ports (Internal Call Storage flag
and Call Forward Ring Timer) fulfill the feature requirements of the Magneto ports, by opening the
'Station Advanced Feature Template' link.
If the default template fulfills your requirement, retain the default Station Basic Feature Template 01 for
the Magneto ports.
However, if you want to change any of the feature settings in for the Magneto Ports, you may prepare a
different Template104, for example, Template 05 and apply it on the Magneto Ports. To do this,
• Click the 'Station Advanced Feature Template' link to open the page.
• Enter the number of the Template you customized, Template 05 in the 'Station Advanced Feature
Template' field of the Magneto Port, for example, MAG-001, on which you want to apply this
template. If you want to apply this template to other terminals too, like MAG-002, 003, and 004,
assign the Template 05 to all these ports.
• Repeat the same steps to customize and assign a different Template to another Magneto Port.
Also refer the topic “Station Advanced Feature Template” for instructions on customizing these
templates and applying them on the station ports.
104. This is recommended because changing the values of the default Template will be applied on all other station types to which the
Template is assigned.
• Now, configure the Magneto Ring Enable (MRE) Key for DKP extensions.
Any DSS key may be configured as MRE Key. There are two ways to do this:
• Assign the MRE Key function in the DKP Key Template assigned to the DKPs, by customizing a template
and assigning it to the DKPs.
OR
• Assign the MRE Key function individually in each DKP, by selecting 'Personalized' Key map for each DKP.
• Go to 'Enable Silence Detection on Magneto'. By default the flag is enabled. If not, select check box to
enable this flag.
• Go to 'Magneto Silence Disconnection Timer', and set it to the desired value. The range of this timer is
from 001 to 255 seconds. By default it is set to 60 seconds.
Key Number is the number of the DSS/Feature Key on the DKP which is to be assigned the MRE
function according to the Model of EON in use:
1 to 25 for EONSOFT
1 to 25 for EON42
1 to 29 for EON48
For example, to program MRE feature on Key Number 04, of the Operator's DKP which is EON48: dial
1261-1-1-3-04-18-77
To know more about assigning Features to DKP keys, refer the topic “DSS Keys Programming”.
• Exit SE mode.
This feature is supported on the SLT/TWT-Magneto card of ETERNITY ME only. You can 'swap' the position of two
ports, SLT/TWT and Magneto in the same RJ45 MDF connector on the card.
You can swap the position of ports of a selected connector or of all connectors on the card.
Illustrated below are the pin connections of the SLT, TWT and Magneto ports in the Normal and Swapped Modes.
SLT8-Magneto8 Card
Port Swapping can be done from the System Administrator (SA) mode using Jeeves or dialing SA commands from
a telephone.
Before you swap an SLT/TWT port and a Magneto port, disconnect the SLT/TWT and the Magneto
Telephone connected to these two ports, and connect them according to the pin position in the Swapped
Mode. Refer to the above illustrations of the pin details of the SLT/TWT-Magneto Cards in the 'Normal' and
'Swapped' mode.
• To swap ports on the SLT-Magneto Card, click the 'SLT8-MAG8' link to open the page.
• To swap ports of the TWT-Magneto Card, click the 'TWT8-MAG8' link to open the page.
• The card connector details will appear according to the Slot Number in which the card is installed. Click the
link of the Slot Number of the card for which you want to use port swapping.
• The connector details of the selected card will appear on the page in the normal mode. The page displays
each connector on the card, with the wire-pair colors, the connection status, and the port name you have
programmed for that port.
• Go to the connector number for which you want to use port swapping.
• Select the 'Swap' check box to enable port swapping for the connector.
• When you finish enabling port swapping for the desired connectors on the card, click 'Submit' at the bottom
of the page.
• The page will be refreshed and the connector details of the 'Swapped' mode will appear on the page.
• If you want to restore 'Normal' mode, clear the 'Swap' check box.
• Repeat the same steps if you want to use port swapping for another SLT/TWT-Magneto Card, installed in
the ETERNITY.
Position is
1 for Normal
2 for Swapped
For example, if you want to swap the 3rd port of the TWT8 - MAG8 card inserted in the 1st slot, to dial
1072-046-01-03-2.
If you want to set the 4th port of the SLT8-MAG8 card inserted in 16th slot back to "Normal" position
then user has to dial 1072-046-16-04-1.
• Exit SA mode.
The ETERNITY ME supports a maximum of 32 Loop Dial (LD) trunk ports105. Before you begin configuration of the
LD trunk ports, ensure that the LD combination card (ETERNITY ME SLT8+MAG2+TWT2+LD2+ENM2) has been
installed correctly.
You may configure the LD ports from Jeeves and using a Telephone.
As the LD Port has a dual personality, behaving as a TWT Trunk port for outgoing calls and as an SLT port for
incoming calls. So, the Loop Dial parameters to be configured as a combination of SLT and TWT hardware and
feature parameters.
• LD Port No.: This non-editable field is the number of the software port assigned to the LD port.
• Hardware Slot-Port: 'Slot' is the number of the Universal Slot in which the LD port Card has been
inserted. 'Port' is the number of the LD port on that card.
By default, the ETERNITY will automatically detect and assign the hardware slot and port numbers to
the LD (software) ports. However, if required, you may change the Hardware Slot and Port assigned to
the LD software port.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
• Enable Port: This flag is for enabling or disabling a LD trunk port. When an LD trunk port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clearing the check
box.
• Name: You may assign a 'Name' to each LD trunk port to facilitate identification. Whenever there is an
incoming call on this port, the Name you have programmed will be displayed on the landing extension.
Each trunk of the ETERNITY can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever there are incoming calls on multiple trunks, the
105. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
By default, the Priority of all LD trunks is set to '5'106. Decide what Priority Level you will assign to the
LD trunk port and set the desired level for the trunk port.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the LD Trunk port, as
the port functions as a station for incoming calls.
By default, Station Basic Feature Template 01 is assigned to all stations of the system (SLT ports, DKP
ports, ISDN Terminals, and E&M Lines with Station as Orientation Type).
Check if the default template fulfills the feature requirements (like “Class of Service (COS)”, “Toll
Control”, “OG Trunk Bundle Group”, etc.) of the LD Port when functioning as an SLT.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all LD trunk ports functioning as SLTs, retain Template 01.
If not, customize a Station Basic Feature Templates and assign it to the LD trunk port. To do this,
• Click the 'Station Basic Feature Template' link to open the page.
• Customize Template number 11 and click 'Submit' at the bottom of the page.
• Enter the number of the Template you customized, Template 11 in the 'Station Basic Feature
Template' field of the LD Port, for example, LD-01, on which you want to apply this template.
• Repeat the same steps to customize and assign a different Template to another LD trunk port.
Also, refer the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on the ports.
• SLT Hardware Template: Assign an “SLT Hardware Template” to the LD Trunk port. By assigning an
SLT Hardware Template the LD port will be configured with hardware features of SLTs like AC
Impedance, Answer Signaling type, Speech Transmit and Receive Gains, Open Loop Disconnect,
Loop Current, and Fax connectivity.
There are 50 SLT Hardware Templates that can be customized and assigned to the LD ports. By
default SLT Hardware Template Number 01 is assigned to all the SLTs of the system.
Check if the values in this template fulfill your requirements. If the default SLT Hardware Template 01
fulfills the feature requirements and if the same features are to be allowed to all LD trunk ports, retain
Template 01.
• Customize Template number 03 and click 'Submit' at the bottom of the page.
• Enter the number of the Template you customized, Template 03 in the 'SLT Hardware Template'
field of the LD Ports on which you want to apply this template.
• Repeat the same steps to customize and assign a different SLT Hardware Template to another LD
port.
Also, refer the topic “SLT Hardware Template” to know more about customizing the templates and
applying on the LD trunk ports.
• TWT Hardware Template: Apply a “TWT Hardware Template” to the LD trunk port. By assigning a
hardware template, you can configure the hardware features for the LD trunk port, such as Trunk Type,
AC Termination Impedance, Pulse-Tone Dialing, Answer Supervision, Disconnect Supervision, DTMF
detection, etc.
The ETERNITY offers 50 TWT Hardware Templates. By default, TWT Hardware Template number 01
is assigned to all TWT Trunks of the system.
Check if this default template fulfills the feature requirements of the LD trunk ports by opening the 'TWT
Hardware Template' link.
If TWT Hardware Template 01 fulfills your requirements, and if the same features at to be applied on all
LD trunk ports, retain Template 01. Similarly, if you want only a few changes to be made to Template
01 and apply it on all LD as well as other TWT Ports, make the changes and retain the template.
However, if different sets of features are to be allowed to different LD trunk ports, then prepare
separate TWT Hardware Templates and apply them on the ports as required. To do this,
• Apply the TWT Hardware Template you customized to the TWT Port by entering the template
number in the 'TWT Hardware Template' field of this port.
• Repeat the same steps to customize another template and apply it to the LD trunk port.
To know more about the hardware port features and customizing templates, refer the topic “TWT
Hardware Template”.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the LD
trunk port for outgoing calls.
Cost Factor is used to grade the cost of routing calls from an LD (TWT) trunk, from 1 to 99; where 1
denotes least cost and 99 denotes highest cost.
Assign a Cost Factor to the LD trunk port, for example, 02 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the LD trunk port 01
only,
• You must first assign a Cost Factor (01-99) to LD trunk port 02, for example, 02.
• Click the 'Least Cost Routing - Number Based' link to open the page.
• Enter '6' in the 'Number' column, Cost Factor '02' as Preference 1, 2, 3 and 4.
All outgoing calls assigned Cost Factor trunk 02 will be made from LD trunk port 02.
• SMDR-OG Storage: This flag is used to enable or disable the storage of details of outgoing calls from
the LD trunk port. Please refer the topic “Station Message Detail Recording-Storage” for more details.
By default, storage of outgoing calls is enabled.
• SMDR IC Storage: This flag is used to enable or disable storage of details of incoming calls on the LD
trunk port. Please refer the topic “Station Message Detail Recording-Storage” to know more. By
default, storage of incoming calls is enabled.
• Hold on DSS Key Press: This flag defines the 'Hold' state of the external called party, when an
extension user presses a DSS key to dial another port.
For example, the DKP extension user (on DKP-001 port) is in the middle of speech with an external
party on LD trunk port-02.
If extension user of DKP-001 presses a DSS key to call another extension port DKP-003, two situations
are possible, depending on whether the Hold on DSS Key Press flag is enabled or disabled:
• When the Hold Flag is enabled: LD-02 will be played music-on-hold. DKP-001 will hear Ring Back
Tone and the call will be placed on DKP-003.
• When Hold Flag is disabled: LD-02 will be disconnected. DKP-001 will hear Ring Back Tone, and
call will be placed on DKP-003.
• Call Cost Calculation Pulse Rate Option: This parameter is to be configured only if you want to apply
the “Call Cost Calculation (CCC)” feature on the LD trunk ports.
You can program four options for Pulse Rate Types. Select from Pulse Rate Type for Pulse Rate
Option 1 to 4 which you want to apply on the LD trunk ports.
• Call Cost Calculation Time Schedule: This parameter is to be configured only if you want to apply the
“Call Cost Calculation (CCC)” feature on the LD trunk ports.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases,
you must first define the Time Zone (time of the day) for which a particular Pulse Rate should be
applied and the Time Schedule for each Time Zone.
You can configure up to four different Time Zones - T1, T2, T3 and T4 with different Pulse rates in the
“Holiday Pulse Rate Table”.
Now, configure the Call Cost Calculation Time Schedule, by specifying the Start Time and the End time
(in 24hours: minutes format) for each Time Zone.
The default Time Schedule (starts and end time) for each Time Zone Index are as follows:
T1 00:00 23:59
T2 00:00 23:59
T3 00:00 23:59
T4 00:00 23:59
If your service provider offers the same Pulse Rate for the entire day,
• program only one Time Zone Index with the Pulse Rate, for example, T1, in the CC-Holiday Pulse
Rate Table.
• Now program the Time Schedule for Time Zone, T1, with the start and end time in Hours: Minutes
format;
• set the start and end time of the other Time Zone Index, T2 to T4, to 00:00 (hours: minutes).
Similarly, if your service provider supports two different Pulse Rates in a day, program the Start and the
End time for two Time Zones and set the other two to 00:00.
• If you have programmed all the Parameters, click 'Submit' at the bottom of the page to save your
settings.
• When you have finished configuring the desired number of LD ports, you may log out of Jeeves or
continue with other configuration tasks.
Flag is:
0 for Disable
1 for Enable
To assign Station Basic Features Template (SBFT) for LD trunk port, dial:
• 5512-1-LD-SBFT to assign a template to a single port.
• 5512-2-LD-LD Trunk-SBFT to assign the same template to a range of ports.
• 5512-*-SBFT to assign the same template to all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
SBFT is from 01 to 50.
By default, SBFT 01 is assigned.
To enable/disable the Hold on DSS Key Press flag for LD trunk port, dial:
• 3945-1-LD-Flag to enable/disable flag for a single port.
• 3945-2-LD-LD-Flag to enable/disable flag for a range of ports.
• 3945-*-Flag to enable/disable flag for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
To program the Call Cost Calculation Pulse Rate Option for LD trunk port, dial:
• 3946-1-LD-Option to program pulse rate option for a single port.
• 3946-2-LD-LD-Option to program the same pulse rate option for a range of ports.
• 3946-*-Option to program the same pulse rate option for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
Option is from 1 to 4.
By default, Pulse Rate Option is set to 1.
To program the Call Cost Calculation Time Schedule-T1-Start Time for LD trunk port, dial:
• 3947-1-LD-Start Time to program start time for a single port.
• 3947-2-LD-LD-Start Time to program the same start time for a range of ports.
• 3947-*-Start Time to program the same start time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The Start Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, Start Time is set to 00:00.
To program the Call Cost Calculation Time Schedule-T1-End Time for LD trunk port, dial:
• 3948-1-LD-End Time to program end time for a single port.
• 3948-2-LD-LD-End Time to program the same end time for a range of ports.
• 3948-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
To program the Call Cost Calculation Time Schedule-T2-Start Time for LD trunk port, dial:
• 3949-1-LD-Start Time to program the start time for a single port.
• 3949-2-LD-LD-Start Time to program the same start time for a range of ports.
• 3949-*-Start Time to program the same start time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The Start Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, Start Time is set to 00:00.
To program the Call Cost Calculation Time Schedule-T2-End Time for LD trunk port, dial:
• 3950-1-LD-End Time to program the end time for a single port.
• 3950-2-LD-LD-End Time to program the same end time for a range of ports.
• 3950-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
To program the Call Cost Calculation Time Schedule-T3-Start Time for LD trunk port, dial:
• 3951-1-LD Trunk-Start Time to program the start time for a single port.
• 3951-2-LD Trunk-LD Trunk-Start Time to program the same start time for a range of ports.
To program the Call Cost Calculation Time Schedule-T3-End Time for LD trunk port, dial:
• 3952-1-LD-End Time to program end time for a single port.
• 3952-2-LD-LD-End Time to program the same end time for a range of ports.
• 3952-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
To program the Call Cost Calculation Time Schedule-T4-Start Time for LD trunk port, dial:
• 3953-1-LD-Start Time to program start time for a single port.
• 3953-2-LD-LD-Start Time to program the same start time for a range of ports.
• 3953-*-Start Time to program the same start time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The Start Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, Start Time is set to 00:00.
To program the Call Cost Calculation Time Schedule-T4-End Time for LD trunk port, dial:
• 3954-1-LD-End Time to program end time for a single port.
• 3954-2-LD-LD-End Time to program the same end time for a range of ports.
• 3954-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
• Exit SE mode.
• Installed the VMS Card correctly and observed the Reset Cycle.
• Changed the IP Address of the VMS Ethernet Port so that it does not conflict with the IP Address of any
other device on the LAN.
• Changed the Subnet Mask of the VMS Ethernet Port so that the port is in the same Subnet as other
devices on the LAN.
For instructions, refer the separate the VMS Card System Manual.
Now,
1. Open the browser Mozilla Firefox or Internet Explorer on the standalone PC or grab any PC on the LAN
and open the browser.
2. Enter the IP Address of the VMS Card Ethernet Port on the address bar of your browser.
4. Configure the VMS Card referring to the instructions given in the separate the VMS Card System
Manual.
Least Cost Routing (also referred to as Automatic Route Selection) is an expense control feature of ETERNITY.
Least Cost Routing (LCR) is useful when there are different trunk lines for making outgoing calls, and the service
providers of these trunks offer different tariffs for calls made to certain locations or numbers or during a particular
time of the day.
When a call is made from an extension of the ETERNITY, LCR recognizes where the call is going to. It selects the
lowest cost trunk from among all the trunks allotted to that extension to make outgoing calls, depending upon how
the LCR is programmed.
The system can be programmed to select the most cost effective trunk for the time of the day when the call is made
from the extension or to select the most cost effective trunk for the destination number dialed from the extension or
to select the most cost effective trunk considering both time of the day and destination number.
Accordingly, ETERNITY supports four types of LCR which can be programmed, namely:
1. Time-based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers a different tariff according to the time of the day.
For example, Service Provider 1 offers a lower tariff for calls made between 9am to 8pm, while Service
Provider 2 offers a lower tariff for calls made between 8pm to 9am.
When Time-based LCR is programmed, the system uses the Online-dialing logic, whereby digits dialed by
the user are directly passed on to the trunk.
2. Number based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers different tariffs according to the area or distance, or phone numbers dialed. For
instance, Service Provider 1 provides cheaper calling rates for calls made from City A to City B, than
Service Provider 2 and Service Provider 3.
3. Time and Number based LCR: This type of LCR is a combination of number and time based LCR, i.e. the
service providers offer different tariffs according to the time of the day as well as area/distance.
For example, Service Provider 1 offers lower rates for calls made from City A to City B during peak hours
9am to 8pm, as compared to Service Provider 2, whereas Service Provider 2 offers cheaper rates for calls
made from City A to City B during off peak hours (8pm to 9 am).
When Time+Number-based LCR is programmed, the system uses Store and Forward dialing logic,
whereby digits dialed by the user are first stored at a memory location in the system, and then dialed out
on the lowest cost trunk.
4. Service Provider-based LCR: This type of LCR may be used when the same Service Providers offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network. For example, Service Provider 1 offers lower rates to call a Service
Provider1 number in City A and in City B, than for calling numbers of Service Provider 2 in the same cities.
This type of LCR may also be used when the same Service Providers apply different charges for different
subscriber services provided by them. For example, Service Provider 1 offers both Fixed Line as well as
GSM services and applies different charges for fixed line and GSM services.
ETERNITY also supports LCR based on Carrier Pre-Selection. This type of LCR is useful where there exist
different service providers for local and long distance calls. Refer the topic “Least Cost Routing-Carrier
Pre-Selection” to know more.
Cost Factor
For LCR to work, all trunks that are allotted to extensions for making outgoing calls, must first be assigned a Cost
Factor.
Cost factor is used for grading trunks in the order of increasing cost of routing calls, from 01 to 99, where 01
signifies least cost and 99 signifies the highest cost. Thus you can grade up to 99 trunks according to the increasing
cost of routing calls.
After assigning Cost Factor to Trunks, you must configure the Type of LCR to be used on Trunks in the Outgoing
Trunk Bundle Group (OGTBG) allotted to the extensions for making calls.
• Make a table of the trunk types and assign a cost factor to each trunk type, as shown below.
TWT-001 BSNL 01
TWT-002 BSNL 02
MOB-001 Reliance 03
MOB-002 BSNL 04
MOB-003 Airtel 05
MOB-004 Vodafone 06
BRI-01 BSNL 07
BRI-02 Reliance 08
• Program the Cost Factor number you assigned to the Trunk types in their respective trunk parameters. For
instance, assign Cost Factor 01 to TWT-001 and Cost Factor 002 to TWT-002 in the "TWT Trunk
Parameters". Similarly, assign Cost Factor 03 to Mobile Trunk 001, Cost factor 04 to Mobile Trunk 002,
Cost Factor 05 to Mobile Trunk port 003, and Cost Factor 06 to Mobile Trunk port 004 in the “Mobile Port
Parameters”.
• For programming instructions, refer the topics “Configuring TWT Trunks” and “Configuring Mobile Trunks”.
• Define the Time Zone, i.e. the start and end time, when the LCR should be applied for the outgoing calls.
The Time Zone you define is stored at an Index number from 1 to 8.
• For each Time Zone that you define, select the Trunk with the lowest cost as your first preference, i.e.
Preference 1. Select the trunk of your second, third and fourth preference (in order of increasing cost.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you have set that is free.
• Refer to the table you prepared for assigning Cost Factor to trunks.
• For example, you want calls made during 9am to 8pm to be routed through BSNL TWT trunks (TWT-001
and TWT-002). If these trunks are busy, you want the system to route calls through the BRI line of BSNL
trunk. When this line is busy, you want the system to attempt to route calls through the BRI line of
Reliance.
• You want calls made between 8pm to 9am to be routed through BSNL TWT trunk 001 only.
• At Time Zone Index 1, define the Time Zone start and end time in 24 Hours:Minutes format, enter the Cost
Factor you assigned to TWT-001 (01) and TWT-002 (02) as Preference 1 and Preference 2 respectively.
Enter the cost factor you assigned to BRI-01 (07) and BRI02 (08) as Preference 3 and Preference 4
respectively.
1 09:00 20:00 01 02 07 08
2 20:01 08:59 01 01 01 01
• Similarly, at Time Zone Index 2, define the Time Zone in 24 Hours: Minutes format. Enter the Cost factor
you assigned to TWT-001, i.e. 01 as Preference 1, 2, 3, and 4. When calls are made during this time
period, they will be routed through TWT-001 only.
• If you have finished defining Time Zones and the preferred trunks for the time zones, configure the Time-
based LCR using Jeeves or a Telephone.
• Click the Least Cost Routing (LCR) link. The links to the LCR options appear.
• Enter the values of the Time-based LCR you prepared on the sheet of paper in the appropriate fields.
To program the Cost Factor (Service Provider preference) for the Time Zone, dial:
• 3403-Time Zone Index-CF1-CF2-CF3-CF4
Where,
Time Zone Index is from 01 to 08.
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.
CF4 is the fourth preferred service provider.
For example, to program the preferred trunks for the Time Zone 09:00 to 20:00 hours, dial 3403-01-01-
02-07-08
It is mandatory to complete this command with CF1 to CF4. If you have only one service provider, program
the same as CF1, CF2, CF3, CF4.
• Exit SE mode.
• Enter each of the number strings at an Index number from 01 to 99. A Number string may be a complete
telephone number, a truncated phone number or an area code.
• For each number string you enter, select the Trunk with the lowest cost as your first preference, i.e.
Preference 1. Select the trunk of your second, third and fourth preference (in order of increasing cost.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you have set that is free.
• Refer to the table you prepared for assigning Cost Factor to trunks.
For example, you want all mobile numbers to be routed through the Mobile Trunk ports, all local numbers
to be routed through the TWT ports.
All mobile numbers start with the number '9', which is prefixed with a '0' when making long distance mobile
calls, so enter '9' and '09' as the number strings. For '9' as well as '09', select the Mobile trunks through
which the calls should be made in order of preference.
Cost Factor
Index Number
Preference 1 Preference 2 Preference 3 Preference 4
1 9 04 03 05 06
2 09 04 06 05 03
3 2 01 02 01 02
99
• If you have finished entering the number strings, and selecting the preferred trunks for the numbers,
configure the Number-based LCR using Jeeves or a Telephone.
• Click the Least Cost Routing (LCR) link. The links to the LCR options appear.
To program Cost Factor (Service Provider preference) for the each Number, dial:
• 3412-Number Index-CF1-CF2-CF3-CF4
Where,
Number Index is from 01 to 99.
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.
CF4 is the fourth preferred service provider.
For example, to program Cost Factor for number '9' at Index 01, dial 3412-01-04-03-05-06
It is mandatory to complete this command with CF1 to CF4. If you have only one service provider, program
the same as CF1, CF2, CF3, CF4.
• Define the Time Zones when the service providers offer lower tariff. You can define up to 8 time zones.
• For each Time Zone you define, specify the Number strings on which lower tariff is applied during that
Time Zone.
• You can enter up to 99 different number strings, which are stored at Index numbers from 01 to 99. The
Number strings may be complete telephone numbers, truncated phone numbers or area codes.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you set as preference if it is free.
For example, service provider of TWT-001 and TWT-002 (assigned Cost Factor 01 and 02) offers the
lowest rate for calls made to Area Code 022 between 8am to 12pm, followed by service providers of
Mobile Trunk-02 (assigned cost factor 04) and Mobile Trunk-01 (assigned cost factor 03).
• Assign Cost Factor preference for the number string in this sequence: 01, 02, 04, 03
Number Index
• If you have finished defining the time zones, entering the number strings, and selecting the preferred
trunks for the number strings, configure the Number and Time-based LCR using Jeeves or a Telephone.
• Click the Least Cost Routing (LCR) link. The links to the different LCR options appear below this link.
• Enter the values of the Time+Number-based LCR you prepared on the sheet of paper in the appropriate
fields.
For example, to define 08:00 to 12:00 as start and end time of Time Zone 1, dial 3421-1-0800-1200
For example, to program Number string '022' at Number Index 01, dial 3422-01-022-#*
To program Cost Factor (Service Provider preference) for the each Number and Time Zone, dial:
• 3423-Number Index-Time Zone Index-CF1-CF2-CF3-CF4
Where,
Number Index is from 01 to 99.
Time Zone Index is from 01 to 08.
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.
CF4 is the fourth preferred service provider.
For example, to program Cost factor for Area code 022 during time zone 08 to 12:00 hours as entered
in the sample table, dial 3423-01-1-01-02-04-03
01 3 080 3 08 07 01 02
02 6 022 3 07 01 02 08
: : :
99 2 03852 5 01 02 01 02
• As you can see, the Service Provider-based LCR Table is similar to the Number-based LCR table.
• You can program as many as 99 different numbers which are stored against Index numbers from 01 to 99.
• The number strings may be the complete telephone number, a truncated phone number or the first digit of
the phone number.
• The Ignore Digit Count is the number of digits in the area code that the system should ignore before
checking the Service Provider-based LCR table. For each area code that you enter, the corresponding
Ignore Digit Count will be the number of digits in the area code. For example, the area code for the number
starting with '3' is 080, which consists of 3 digits. So, the Ignore Digit Count for the number/area code 080
will be 3.
• For each number string and area code that you enter, assign the Trunk of the service provider who offers
the lowest tariff to that number/area code. Refer the table you prepared for assigning Cost Factor to trunks.
• If you have finished entering the number strings, their corresponding area codes and the Ignore Digit
Count, and the preferred trunks, configure Service Provider-based LCR using Jeeves or a Telephone.
• To program Area Code and Ignore Digit Count, click the 'Call Cost Calculation' link.
• Now, click the Least Cost Routing (LCR) link. The links for the LCR options appear.
• Enter the values of the Service Provider-based LCR you prepared on the sheet of paper in the appropriate
fields.
For example, to program Number string '080' at Number Index 001, dial 2620-001-080-#*
Refer the topic "Area Code Table" under Call Cost Calculation to know more.
To program a number at number Index in the Service Provider-based LCR table, dial:
• 3441-Number Index-Number String-#*
Where,
Number Index is from 01 to 99.
Number String can be a complete telephone number, a truncated telephone number or an area code.
Number string may be a maximum of 16 digits. Terminate the command with #* if the number string
has fewer than 16 digits.
By default, Number String is 'Blank'.
To program Cost Factor (Service Provider preference) for the each Number, dial:
• 3442-Number Index-CF1-CF2-CF3-CF4
Where,
Number Index is from 01 to 99.
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.
CF4 is the fourth preferred service provider.
For example, to program Cost Factor for number '3' at Index 01, dial 3412-01-08-07-01-02
It is mandatory to complete this command with CF1 to CF4. If you have only one service provider, program
the same as CF1, CF2, CF3, CF4.
• Exit SE mode.
• Click the 'Outgoing Trunk Bundle Group' link to open the page.
• For each OGTBG number assigned to extensions, select the desired LCR Type: Time-based, Number-
based, Time+Number based, Service Provider-based (Cost Factor).
You can find the OGTBG number assigned to each extension from the Station Basic Feature Template
assigned to the extension.
• Exit SE mode.
The system will disregard features such as Toll Control, Call Budget, Automatic Number Translation, Call Duration
Control on the extensions when dialing out Emergency Numbers. To know more, refer the topic “Emergency
Dialing”.
For Emergency Number Dialing, you need to configure the Emergency Number Table. In this table, each
Emergency Number is to be assigned an Outgoing Trunk Bundle Groups (OGTB) through which the Emergency
Number is to be routed.
The system loads the default Emergency Numbers and Outgoing Trunk Bundle Group (OGTBG) in the Emergency
Number Table as per the “Region” you selected for the system. The Emergency Numbers loaded by default in the
Emergency Number Table are non-editable, but you can re-assign the default OGTB, as per your requirement.
For example, if you selected USA as Region, the default Emergency Number Table would look like this:
01 911 32
02 112 01
03 01
04 01
05 01
06 01
07 01
08 01
09 01
10 01
If you selected Australia as Region, the default Emergency Number Table would look like this:
01 000 32
02 106 30
03 112 31
04 01
05 01
06 01
07 01
08 01
09 01
10 01
Each Number is stored at an index from 01 to 10. The Emergency Number fields from index 01 to 5 are non-
editable, but you can select a different OGTBG for each of these default Emergency Numbers.
You can add Emergency Numbers at index 06 to 10 in the table, and select the OGTBG as required.
The first five entries, at Index 01 to 05 on this table, are uneditable. These fields will be populated with the
default Emergency Numbers of your country (which you selected as Region).
If the Emergency Numbers loaded by default are not applicable for your region/country, you may add the
Emergency Numbers and their OGTBG at index 06 to 10 in this table.
Make sure that the trunks configured in the OGTBG for each Emergency Number belong to the correct
network and the ports through which the calls are to be routed are not disabled. For example, '112' is the
default Emergency Number for the mobile network. So, make sure that the Mobile Trunk to be used for
dialing this number is included in the OGTBG you assign to this number in the Table.
• Exit SE mode.
Abbreviated Dialing
What's this?
Abbreviated Dialing is the use of short codes (abbreviated numbers), typically 2-3 digits, to dial out long-digit
numbers. It is also referred to as Memory Dialing.
Abbreviated Dialing allows you to dial quickly and easily, frequently called, long-digit numbers.
This feature requires you to store the frequently called, long-digit numbers107 and their corresponding short codes
in special lists, known as 'directories'. These directories may be 'personal' or 'global'.
ETERNITY supports two types of Abbreviated Dialing based on the type of directory used: Personal Abbreviated
Dialing and Global Abbreviated Dialing.
Abbreviated Dialing forms the basis of two other features of the ETERNITY: “Dialed Number Directory” and “Quick
Dial”.
Personal Directories can be programmed and assigned to groups of extensions. The use of Personal Directories is
limited to the extensions to which they are assigned.
A personal directory accommodates 25 numbers. Each number may be up to 16 digits long. A personal directory
has Index numbers from 01 to 25 against which the frequently dialed telephone numbers are stored along with their
corresponding names and trunk access code ID.
As many as 50 different personal directories, numbered from 01 to 50 can be created and assigned to SLT, DKP
and SIP extensions.
With a personal directory assigned to an extension, the extension user simply dials out the Feature Access Code
for Abbreviated Dialing and the Index Number at which the desired number is stored in the personal directory.
107.These may be numbers of your branch offices, your clients, as also numbers of emergency services such as fire, police.
ETERNITY will automatically dial out the number using the trunk access code ID specified for this number in the
personal directory.
• When an extension user dials an abbreviated number from the Personal Directory, the system first
checks OG Trunk Bundle Group (OGTBG) and Toll Control Level (Call Privilege) of that extension and
then dials out the number.
• Each extension can access only the personal directory assigned to it.
• Personal Directory can be programmed by the System Engineer, as well as extension users. Extension
users can add contacts to the Personal Directory assigned to them from their extensions phones (DKP/
SLT/ISDN phone/IP Phone).
Being a system-wide list, the Global Directory can be accessed by any extension connected to the ETERNITY.
The Global Directory has the capacity to store up to 900 numbers of a maximum of 16 digits each. The Global
Directory is divided into three parts:
The Global Directory has Memory Location codes starting from 100 to 999. The telephone numbers along with their
corresponding names are stored against Memory Location codes.
Whenever extension users of ETERNITY want to use Global Abbreviated Dialing, all they needs to do is dial the
feature access code ('8' or '6') and the Memory Location code at which the desired number is stored.
For example: the number 02652630566 is stored at Memory Location 102 of the Global Directory. Now, extension
users of ETERNITY can call this number by simply dialing the '8' or '6' (feature access code for Abbreviated
Dialing) followed by '102' (Memory Location code at which the desired number, 02652630566, is stored).
The ETERNITY will dial out the number using any of the trunks in the “OG Trunk Bundle Group” assigned to it in the
Memory Location.
• Extensions can use Global Abbreviated Dialing only if this feature is included in the “Class of Service
(COS)” allowed to them.
• So, to be able to access the entire Global Directory, extensions must be assigned all three parts of the
directory in their Class of Service. By default, only Global Directory Part 1 is included in the CoS of all
extensions.
• Global Directory can be programmed by the System Engineer, and by Digital Keyphone extension
users who have Global Directory Programming allowed to them in the Class of Service.
• While the System Engineer can program all three parts of the Global Directory, digital keyphone
extension users who are allowed Global Directory Programming in their Class of Service can configure
only Global Directory Part 1.
How to configure
For both Personal and Global Abbreviated Dialing to work, the System Engineer must:
• Assign Personal Directory to the desired extensions (which may be different: SLT, DKP, ISDN Terminals,
IP Phones).
• Enable Global Directory Part 1/2/3 as desired in the Class of Service (CoS) group allowed to the
extensions.
• Enable ‘Global Directory Part 1’ and ‘Global Directory Programming’ in the CoS group of the digital key
phone extension users, who are to be allowed to program (add, delete, edit) contacts in the Global
Directory Part 1 from their digital keyphones.
All the above parameters can be programmed by the System Engineer using Jeeves as well as a telephone.
• Ask the extension users the numbers they would like to be included in the personal directory of their
extension.
• Make separate lists of numbers along with their corresponding names and trunk access codes, for each
personal directory. You may draw four-column tables on paper and enter the Numbers and corresponding
names and trunk access codes against each Index number. For example:
: : : :
: : : :
25
Personal Directory 02
: : : :
: : : :
25
• Compile the numbers to be included in the global directory. Numbers that are commonly dialed by all
extensions can be included in the global directory.
• Draw a four-column table on paper and enter the telephone numbers along with their names, and the
Outgoing Trunk Bundle Group (OGTBG) at each Memory location. For example:
Global Directory
100
101
: : : :
: : : :
999
• Prepare the Global Directory keeping in mind that is divided into three parts: Part 1 (100 to 799), Part 2
(800 to 899), and Part 3 (900 to 999). As Part 1 is allowed to all extensions in their default CoS, you may
include the numbers allowed to all extensions in this part of the directory.
• For each Personal Directory, enter the Number you wish to store against an Index Number. Enter the
contact's Name against the number.
The length of the Number field is limited to 16 digits. The length of the Name field is limited to 12
alphanumeric characters. Ensure that the number and the name are programmed within this limit.
Each directory has a limit of 25 entries. You may enter up to 25 Numbers and Names in each Personal
Directory.
Keep a print of each personal directory for your record and for the record of the extension user to whose
phone the personal directory is assigned. This will also help you take care of overlaps and include some of
the numbers that are dialed by all users in the Global Directory instead of the Personal Directory.
After you have programmed the Personal Directories, assign a personal directory to each extensions.
• Enter the number of the Personal Directory. For example, to assign Personal Directory No. 02 to SLT 2001
(software port 001, connected on hardware slot 03, hardware port 09), enter '02' in the 'Personal Directory'
column for SLT 2001.
• Enter the number of the Personal Directory. For example, to assign Personal Directory No. 01 to DKP
3001 (software port 001, connected on hardware slot 17, hardware port 01), enter '01' in the 'Personal
Directory' column for DKP 3001.
• Go to 'Personal Directory' column of the ISDN Terminal to which the directory is to be assigned.
• For each ISDN Terminal that is to be assigned a Personal Directory, enter the number of the directory in
this column.
• Under ‘VoIP Configuration’, click the 'SIP Extension Settings' link to open the page.
• Select the software port number of the SIP extension you want to assign the Personal Directory.
• Exit SE mode.
• Click the 'Global Directory' link under Abbreviated Dialing to open the page.
Each page of has 99 entries. To go to the next 99 entries click the links above the table '200-299'
• Enter the Number you wish to store against a Memory Location Code. Enter the contact's Name
against the number.
The length of the Number field is limited to 16 digits. The length of the Name field is limited to 12
alphanumeric characters. Ensure that the number and the name are programmed within this limit.
• Now, apply the Global Directory to extensions. Make sure that the feature Global Directory is enabled
in the CoS of the extensions to which you are assigning the Global Directory.
If the entire directory is to be assigned to all extensions, you may simply enable 'Global Directory Part 2
and Part 3 in the default CoS group 01 in the default Station Basic Feature Template 01 assigned to
the extensions.
However, if selected extensions are to be allowed Global Directory Part 2/Part 3, follow these steps:
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for further instructions.
• Decide which of the DKP Extension users are to be allowed ‘Global Directory Programming’ (of Global
Directory Part 1) and allow this feature in their Class of Service.
By default, ‘Global Directory Programming’ is disabled in the default CoS group 01 in the default
Station Basic Feature Template 01 assigned to all extensions of ETERNITY. This means none of the
extensions can program Global Directory.
If you want to allow Global Directory Programming to all DKP extension users, simply enable this
feature in the CoS group of the Station Basic Feature Template assigned to them.
If you want to allow Global Directory Programming to only selected extensions, then follow these steps:
• Define a CoS group with Global Directory Programming enabled.
• Make sure this CoS also has Global Directory Part 1 enabled.
• Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
• Assign this template to the DKP extensions to which Global Directory Programming is to be
allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for programming
instructions.
• When Global Directory Part 1 is assigned to an extension user, ETERNITY will not check for Toll
Control.
• When you assign Global Directory Programming to a DKP extension user, the user can program any
number in Global Directory Part 1, this includes numbers denied to the extension user in the Call
Privilege defined in the Toll Control level of this extension user.
• Since the system does not check for Toll Control for numbers dialed out from Global Directory Part 1,
there is a possibility of extension users programming numbers not allowed to them in their Toll Control
level in the Global Directory Part 1, inadvertently or intentionally.
• Hence, the System Engineer is advised to exercise caution when allowing this feature to DKP
extension users.
• Click ‘Submit’ at the bottom of the pages on which you make changes to save your settings.
• If you have finished configuration, you may log out of Jeeves. Or you may continue, as required.
• Exit SE mode.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions on how to use
SE commands to
• apply the template with the newly programmed CoS group to the extensions.
If you are using an SLT, you will not be able to program the Name of the contact in the directory.
OR
• Dial 1071.
• Enter Personal Memory Index (001 to 025)
• Enter Number of the contact (max. 16 digits).
• Press 'Enter' key.
• Enter Name of the contact.
• Press 'Enter' key.
• Enter Trunk Access Code.
• Press 'Enter' key.
• You get confirmation tone and the message on your phone's display.
• Extension users can only add, delete and edit names and numbers of contacts in Global Directory Part
1. However, they cannot program the Outgoing Trunk Bundle Group (OGTBG) for the contacts in the
directory.
• When an extension user programs Global Directory Part 1, the system will automatically assign the
number and name to a free Memory Location. The system will use the OGTBG assigned to that
Memory Location by the System Engineer to dial out the number added by the extension user.
• By default, OGTBG 01 is assigned to all Memory Location Codes in the Global Directory.
• If no OGTBG has been assigned to a Memory Location in the Global Directory (i.e. the field is blank),
and an extension user adds a contact to this Memory Location, the number will not be dialed out.
•
To program Global Directory Part 1 from the digital key phone, follow these steps:
Adding a contact
To add a contact, select ‘Add’ and press Enter key.
• Enter your contact’s name on the prompt: ‘Name:’
A maximum of 12 characters are allowed.
• Press Enter key to save name.
• Enter your contact’s number on the prompt: ‘Number:’
A maximum of 16 digits are allowed.
• Press Enter key to save number.
• You will get the confirmation tone and the confirmatory message: “Stored at Index xxx”.
Editing a contact
To edit a contact,
• Scroll to ‘Contacts’ in the phone menu and press Enter key.
• Select ‘Edit’ and press Enter key.
• You get the prompt: 'Name:'
• Enter the initial letters of the contact's name.
• The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
• Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
• Press 'Enter' key to select the name.
• The system displays the name you selected.
• To delete a character, use the Back/Forward navigation key to place the cursor under the character you
want to delete.
• Press the ‘Transfer’ key to delete the character you selected with the cursor.
• To enter a character, use the Back/Forward navigation key to place the cursor in the position you want to
enter the character.
• Enter the desired character by pressing the relevant digit pad keys in quick succession.
• After you have finished editing the name/ number, press Enter key.
• The number of the contact whose name you edited will be displayed.
• Repeat the same steps as you did for editing the name.
• After you have finished editing the number, press Enter key.
• You will get the confirmation tone and the confirmatory message: “Stored at Index xxx”.
How to use
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What's this?
Access codes are short digit sequences dialed from an extension phone to instruct the PBX to perform a function
such as:
• Calling an extension.
• Grabbing a trunk line or any trunk line from a group of trunks (“OG Trunk Bundle Group”).
• Station Codes: Codes used for calling extensions, Analog Input Port, Digital Output Port. These codes
are also commonly referred to extension numbers, phone numbers. For the purpose of this document,
station codes are referred to as Flexible Numbers.
Default station codes: the factory-set default values for SLT extensions are from 2001 to 2512; for DKP
extensions from 3001 to 3128; for Analog Output Ports from 3921, 3922….
• Logical Group Codes: codes used for calling a group of stations as in a Department group, a group of
trunks as in Outgoing Trunk Bundle Group.
Default logical group codes: the factory-set codes for Department Numbers start from 3901, 3902….
Outgoing Trunk Bundle Groups from 61, 62, etc.
Default feature codes: there are different feature codes for every feature/function of the ETERNITY, e.g.: '2'
for Auto Call Back, '5' for Raid, '13' for Call Forward, etc.
You can change the default access codes to codes of your choice. For example: the default Operator code
'9' can be changed to '0'; the default trunk access code for dialing Trunk Group1 can be changed from '61'
to '5'.
How it works
Whenever an access code is dialed from an extension, the system matches each digit in the code with the access
codes programmed within the system to determine the instruction, i.e. whether it is an extension it must call, or a
trunk line it must grab, a port it has to activate, etc. The system processes the instruction when a match is found.
• When the first digit '1' is dialed, the system finds a match. As several default access codes begin with '1'
the system waits for the next digit to be dialed.
• When the second digit '3' is dialed, the system finds a match for '13'.
• As '13' is common for all Call Forward options108, the system waits for the next digit to be dialed
• When the user dials the third digit '1', the system finds a match for '131'.
• If there is more than one access codes matching with '131', e.g. '1311', '1314', '1315' the system will wait
for the next digit to be dialed.
• If no further digit is dialed on expiry of the Inter Digit Wait Timer, the system understands the instruction as
'Call Forward - Unconditional' and waits for the destination phone number to be dialed.
Access Codes are related to various phases of a call. When a call is processed by a PBX, it goes through a
number of pre-defined phases.
No Digits are The system The dialed The dialed Connected Connected No reply
activity. pressed on is processing extension extension is with the with two from dialed
the phone the call. The is busy. ringing. dialed extensions. extension.
keypad/dialed call is neither extension.
from the placed nor
rotary. blocked.
Dial tone is Beeps are Busy tone Ring Back Two-way Three-way Error Tone
played. played. is played. Tone is speech. speech. is played.
played.
Different access codes are dialed at different call phases. Station Codes and Logical Group Codes are dialed in the
'Dial' phase.
As different features are invoked in each call phase, Feature Access Codes are dialed at different call phases. For
example:
• Auto Call Back code is dialed at the 'Blocked' phase' as well as 'Placed' phase.
108. Call forwarding options: Unconditional, When Busy, When No Reply, When Busy or No Reply.
Each access code in a single call phase may be of different lengths, but must be unique. For example, the
same access code cannot be used for two different features like Call Forward and Redial, since both these
features are invoked in the 'Dial' phase.
However, the same access code can be used for features in different call phases. For example, '4' is the
default feature access code for DND Override (Routing Phase), Call Pick-Up-Group (Dial Phase) and Barge-
In (Blocked Phase).
Similarly, Station and Logical Group Codes too must be unique and should not match with any of the features
invoked in the 'Dial' phase. Refer the topics “Flexible Numbers” and “OG Trunk Bundle Group” to know more.
How to configure
ETERNITY provides default Access Codes for stations, logical groups - department and trunk groups - and
features.
It also provides country-specific default Access Codes which are applied automatically when you select the
'Region' to configure the system.
The default Access Codes for India are presented in the table below. The default Access Code tables also indicate
the call phase in which each feature is invoked.
Call Phases
Feature Access
Feature Matured Matured
Number Code Dial Routing Blocked Placed
2-Way 3-Way
Redial 07 7 Y Y
Auto Redial 08 17 Y Y
Operator Dialing 12 9 Y Y Y
Dynamic Lock 14 14 Y Y
Hot Line 15 15 Y Y
Alarm 16 161 Y Y
Do Not Disturb 17 18 Y Y
Interrupt Request 18 3 Y
Barge-In 19 4 Y
Raid 20 5 Y
Trunk Reservation 21 6 Y
Call Toggle 22 1 Y
Conference- 3 Party 23 0 Y
Conference-Multi Party 25 19 Y Y
Flashing on Trunk 38 * Y Y
DND Override 45 4 Y
Forced Release 49 #* Y
Hold 50 Flash Y
Forced Answer 52 5 Y
Mute 56 1052 Y Y
SA Command 60 1072 Y Y
Floor Service 62 38 Y Y
CLIR 64 103 Y
Reminder 66 162 Y Y
RCOC Invoke 78 **
You can either use the default Access Codes or change them to suit your preferences.
If you enter a code that is already assigned to a station or a feature, the system will not accept the
duplicated code. The value will remain unchanged.
• To disable an access code, delete the existing code and leave the field blank.
To change Station Access Codes (for SLT and DKP), Department Groups, Trunks and Trunk Groups, refer
the topics “Flexible Numbers”, “Department Call”, and “OG Trunk Bundle Group”.
If you try to assign a number string that is already used to access a station or use a feature then the
system will not accept the command and will play error tone.
• Exit SE mode.
What's this?
Account Codes are a very useful feature for organizations such as business consultants, law firms, advertising and
media agencies, and the like that cater to several clients, interacting with third parties on behalf of their clients.
Such organizations need to keep track of calls made to and on behalf of each client.
An 'Account Code' is a unique three-digit number that an organization can assign to each of its clients. Each
Account Code may be given a name and programmed in the Account Name List.
Doing so, whenever calls are made to the client or to a third party on behalf of the client,
• The extension user dials the Account Code or Name assigned to the client.
• Details of these calls are recorded by the Account Code dialed in the Station Message Detail Recording
Report (SMDR) for Outgoing Calls.
• The SMDR report can be printed using the Account Code as filter.
This way, the organization can know the details of calls made to and on behalf of each client.
To illustrate this with an example: An advertising media agency makes nearly 100 calls every day to and
on behalf of its clients that include 'Midas Business Solutions', 'Jet-Set Holidays', 'Bacchus Vineyard',
among several others.
Now, the agency can assign a three-digit account code to Midas Business Solutions, for instance '001' and
the name code 'Midas Biz' in the Account Name List. The same can be done for all other clients. Each time
someone in the agency makes a call to Midas, they may dial either the account code '001' or the account
name 'Midas Biz', either before dialing out the number or when in speech with the client.
• an extension user dials out the number or the Trunk Access Code to grab a trunk.
• the ETERNITY displays the Account Name List on the DKP of the extension user.
• To use Account Codes, this feature must be included in the Class of Service (CoS) group allowed to
the extensions.
• If you want to use Account Names, you must program the Account Name List.
• When the Forced Account Code is enabled on an extension and trunk, the system will ask the user to
enter the account code irrespective of the method of dialing: Global Abbreviated dialing, Personal
abbreviated dialing, Least Cost Routing, or Selective Trunk Access.
• However, if Forced Account Code is enabled on the selected trunk, and the number is dialed using
Selective Trunk Access, the system will dial out the number using Store and Forward dialing.
In the case of Abbreviated Dialing or Direct Dialing, if the extension user fails to dial the Account Code,
an error message will be displayed on the extension user's DKP.
How to configure
For Account Code to work, the System Engineer must:
1. Enable 'Account Codes' feature in the Class of Service (CoS) of the extensions to which this feature is to
be allowed.
3. If Forced Account Code is to be used, the System Engineer must enable 'Forced Account Code' flag in
• the “Station Advanced Feature Template” applied to the extensions from which calls using account
codes are to be made.
• the “Trunk Feature Template” applied on the trunks through which calls using account codes are to be
made.
All the above feature parameters can be programmed using Jeeves or by dialing commands from a
Telephone.
• Write Account Codes on one column. Account codes may be any three-digit number between 001 and
999.
010 Bacchus
You need not follow a cardinal numbering sequence when assigning Account Codes.
You may assign any code to any client. For instance, you can assign code '111' to Midas Business
Solutions, '222' to Jet-Set Holidays, '333' to Bacchus Vineyard.
Now, include the feature 'Account Codes' in the Class of Service of the extensions.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of the
ETERNITY. Template 01 has the feature Account Codes in the default CoS Group (Number 01). So, all extensions
of ETERNITY can use this feature.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the stations to which Account Codes is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions.
If Forced Account Code is to be used, enable the 'Forced Account Code' flag in the “Station Advanced Feature
Template” of the stations and in the “Trunk Feature Template” assigned to the trunks.
When you enable 'Forced Account Code' in a template, this feature will be enabled on all DKP, SLT and
SIP extensions that are assigned this template. If necessary, create a separate template with this feature
and assign this template only to those stations that are to be assigned this feature.
• Go to the parameters page of the type of trunk you want to program. For example:
• Click the ‘Trunk Feature Template’ link in the column. The feature template page will open.
If you want to enable this feature on all trunks, enable it in Trunk Feature Template 01.
If you want to enable this feature only on select trunks, program a different Template number with this
feature.
• Select the ‘Forced Account Code' check box in the template number assigned to the trunk.
• Now change the Trunk Feature Template number of the trunk you want to program. This number should
be the same as the template in which you have enabled the Forced Account Code check box.
For example: To enable Forced Account Code Flag in Station Advanced Feature Template 02: Dial
5602-1-02-09-1
To apply the Station Advanced Feature Template now programmed with the Forced Account Code on
stations, refer the topic “Customizing Station Advanced Feature Template using a Telephone”.
For example: To enable Forced Account Code Flag in Trunk Feature Template 01: Dial 5802-1-01-29-1
To apply the Trunk Feature Template now programmed with the Forced Account Code on different
types of trunks, refer the topic “Customizing Trunk Feature Template using a Telephone”.
• Exit SE mode.
Print and hand out copies of the Account Code List to everyone in the organization for reference while
making calls.
OR
• Dial 1058
• Enter Account Code
• Dial Trunk Access Code
• Dial the number of the client.
OR
• Dial 1058
• Enter Account Code
• Speech will be resumed.
To enter Account Code Number when Forced Account Code Flag is enabled:
• Dial 1058
OR
• Dial 1059.
• Enter the initial letter of the client's name.
The Account Name List will be displayed on your DKP, alphabetically with the corresponding account
codes.
• Scroll to select the desired client name and press Enter key.
• Dial Trunk Access Code.
• Dial the client's number.
OR
• Dial 1059.
• Dial the initial letter of the client's name.
The Account Name List will be displayed on your phone, alphabetically with the corresponding account
codes.
• Scroll to select the desired client name and press Enter key.
Speech will be resumed with the called party.
If you have dialed the wrong account code or name while in the middle of a call, you can correct it by
pressing 'Hold' again and following the steps described above. The system will override the previously
dialed account code or name.
Refer the section “Station Message Detail Recording-Report”, for more detailed instructions on printing reports
using filters.
What's this?
Alarms are an efficient and user-friendly feature available to all extensions of the ETERNITY.
• Once Only - A one-time call, where the extension phone rings at the set time.
• Daily - A repeat call, where the extension phone rings at the set time everyday.
• Personalized - The Operator greets the extension user to serve the alarm request.
• Automated - The system serves the alarm request by playing a voice message or music.
How it works
Personalized Alarm
When the Alarm serving mechanism is configured as 'Personalized',
• The Operator phone rings first109, displaying the number of the extension to which the alarm is to be
served.
• When the Operator answers this call, a call is placed on the extension on which the alarm is set.
• The extension rings for the duration of the Alarm Ring Timer.
• When the extension user answers the call, the Operator greets the extension user with the time and alarm
message.
• This event is recorded in the Hotel-Motel Activity Log as 'Wake-up Alarm of <HH:MM> Answered on
<phone number>'.
• If the extension user does not answer the call till the Alarm Ring Timer has elapsed, the Operator phone
will display a text message notifying 'No Reply' from the extension. The Alarm is now considered as
served.
109. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the ETERNITY will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.
• If the extension is busy110, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
• inform the extension user about the alarm in person or send someone to do it.
Automated Alarm
When the Alarm serving mechanism is configured as 'Automated',
• The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is
from the EON series, an Alarm message will appear on its display.
• When the extension user answers the call, s/he may be played music-on-hold, or a pre-recorded voice
message, or a music/message from an external source111, or be connected to a routing group, depending
upon the Alarm Notification Type programmed by the System Engineer.
The System Engineer may consult with the Enterprise to decide which of these options is to be
programmed as the Alarm Notification Type.
• If the extension user does not answer the alarm call, the ETERNITY makes two more attempts (in all, 3
attempts) at an interval of 5 minutes between each attempt, to call the extension. (Each attempt is
recorded in the Hotel-Motel Activity log as 'Wake-up Alarm of <HH:MM> No Reply on <Phone Number>'.
• If all Alarm attempts go unanswered, the ETERNITY places the call on the Operator phone. The Operator
phone rings till the end of the Alarm Ring Timer. The Operator phone displays the extension number with
the message 'No Reply'. The Alarm is now considered as served. (This event is recorded as "Alarm
Notification to Front Desk for <Phone Number>").
• If the extension phone is busy ETERNITY will continue to make Alarm Attempts at the Alarm Interval
programmed. When all Alarm Attempts go unanswered, the ETERNITY will place a call on the Operator
phone. The Operator phone will display the number of the extension phone with the message 'Busy'.
The Snooze function can be added to Automated-Alarms to ensure that the extension user answers the
call. Snooze is a system-wide feature; when set, this function will be added to all Automated Alarms.
110. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'. This event is recorded in the Hotel-Motel Activity Log as "Alarm not Served, <phone
number> is Busy".
111. This device can be connected to the Analog Input Port of ETERNITY.
The status of Alarms set by Operator as well as extension users appears on this report, with details of time (hours
and minutes), type (once only, daily), and serving mechanism (personalized, automated).
The Operator can generate the Alarm report any time, or can set the system to automatically generate the Alarm
Report at a particular time. This is referred to as Scheduled Alarm Report.
The Alarm Report generated by the system can be printed or sent to a computer.
• ETERNITY can register as many as 960 Alarm requests set by the Operator and extension users.
• Multiple Alarms can be set for an extension by the Operator and/or by the extension user. For example,
Daily Alarm at 09:00am is set for an extension. The extension user wants to change the alarm time to
08:30am for a day. The extension user/Operator can set another alarm, i.e. a Once Only Alarm, at
08:30am without disturbing the daily alarm. Both the Alarms will ring at the set time.
• When multiple alarm requests have been set on an extension, if the Operator/extension user cancels
an alarm set for an extension, the system cancels all alarms set for the extension. It is not possible to
cancel any of these alarms selectively.
• It is not possible to modify an alarm request. Instead, the alarm request should be canceled and a new
one should be made.
• The duration of Alarm Ring Timer, the Number of Alarm Attempts and the Alarm Attempt Interval are
programmable.
• Alarms can be set for all extensions of the ETERNITY, including the Operator phone also.
• All the Alarm events are logged in the "Hotel-Motel Activity Log".
• Alarm settings will be retained in the system during power down and system upgrades.
How to configure
The following parameters play an important role in the functioning of the Alarm feature. These parameters carry
default values. The default values have been selected keeping the larger user base in mind. However, these values
can be changed by the System Engineer at the time of installation or afterwards as per users' requirements.
1. Alarm Ring Timer - The duration for which the system rings the extension to serve an Alarm call. By
default, the Alarm Ring Timer is set to 45 seconds. This timer can be set between 001 to 255 seconds.
This timer also signifies the duration for which the Operator phone rings to notify that an Alarm call has not
been answered or the extension phone is busy.
2. Number of Alarm Attempts - Number of times the system attempts to place an Alarm call on the
extension phone before notifying the Operator that the call is not answered or the phone is busy. By
3. Alarm Attempt Interval - The time period between each Alarm Call attempt. By default, the Alarm Attempt
Interval is set to 5 minutes. The Alarm Attempt Interval can be set between 1 and 9.
4. Snooze Functionality - Snooze is a functionality which forces the extension user to acknowledge the
Alarm call. With snooze functionality enabled, the system expects the user to answer the Alarm call by
going OFF-Hook and dial Acknowledgement code '0'. With snooze functionality disabled, the system
considers the Alarm as answered when the extension user simply answers the alarm call by going OFF-
Hook (dialing acknowledgement code is not mandatory). Users may choose whether or not to enable
snooze functionality. By default, snooze functionality is disabled.
5. Configurable Alarm Type - When the Operator and extension user set an Alarm call request, the system
gives them the choice of setting 'Once Only' or 'Daily' Alarm calls.
User experience however, shows that 'Once Only' Alarm call requests are more common than 'Daily' Alarm
requests. So, ETERNITY allows you the flexibility of setting 'Once Only' as the default Alarm Type, by
disabling the 'Configuring Alarm Type' flag.
When this flag is disabled the system will prompt the Operator/Extension user to enter the Time of the
Alarm call and consider the Alarm Type as 'Once Only'.
6. Configurable Alarm Category - When the Operator sets an Alarm call for an extension, the system
prompts the Operator to select an Alarm Type (Once Only or Daily) and to select the alarm serving
mechanism - 'Automated or Personalized'.
If the Enterprise wishes to offer only 'Automated' Alarms to its extension users, ETERNITY allows the
flexibility to set 'Automated' as the default Alarm call serving mechanism. This can be done by disabling
the 'Configurable Alarm Category' flag.
When this flag is disabled, the system will consider the Alarm call serving mechanism as 'Automated' and
will prompt the Operator only for the Time of the Alarm call.
• When both flags 'Configurable Alarm Type' and 'Configurable Alarm Category' are disabled, the system
will set and serve 'Once Only - Automated' alarms only.
• If the 'Configurable Alarm Type' flag is disabled, but the 'Configurable Alarm Category' flag is enabled,
the system will set 'Once Only' alarm calls, but give the option of selecting 'Automated' or
'Personalized' as the serving mechanism.
• Similarly, if 'Configurable Alarm Type' is enabled, but the 'Configurable Alarm Category' flag is disabled,
the system will allow both 'Once Only' and 'Daily' alarms to be set, but the serving mechanism will be
'Automated'.
7. Alarm Notification Type - This is the means of notifying the extension user about the Alarm call. The
extension user can be played Music-On-Hold, Live Music, Pre-recorded Voice Message, Weather
information, Date and Time, etc. The ETERNITY supports four types of Alarm Notifications:
b. Music-On-Hold: Selecting this option would play music-on-hold to the extension user when s/he
answers the Alarm call.
c. External Music Source: Selecting this option would connect the extension user to live music when he
answers the Alarm call.
However, for this option to work the System Engineer should connect a live music source to the Analog
Input Port of ETERNITY. The live music source can be replaced by any other form of music. Please
refer the 'Technical Specifications' of Analog Input Port.
d. Routing Group: Selecting this option would connect the extension user to the stations programmed in
the Alarm Notification Group. The System Engineer may connect a device which can play customized
alarm greetings with date, time, weather conditions, traffic conditions, a marketing message, etc. on the
stations programmed in the Alarm Notification Group.
8. Macros - This is a short code for simulating the Alarm call. The SLTs with special function keys send a
fixed string to the system, when each function key is pressed. The system interprets this string and
translates it into a string that can be understood by the system. For example, the SLT has a special
function key for Alarm calls which sends the string '51' to the system. The system can be programmed to
translate '51' into the feature access code for Alarm calls, '*161'.
All the above listed parameters can be programmed using Jeeves and a Telephone.
• Use Alarm with snooze - enable this flag if you want to use the Snooze function for the Alarm Call.
• Alarm Ring Timer (Sec.) - you may change the time for which the Alarm Call will ring on the extension
phone and the time for which the Operator phone will ring to notify an unanswered Alarm Call.
• Number of Alarm Attempts - you may increase or decrease the number of attempts the system
should make to serve an Alarm call.
• Alarm Attempt Interval - you may increase or decrease the time gap between each attempt the
system makes to serve an Alarm call.
• Configurable Alarm Type flag - disable this flag, if you do not what the system to provide the
Operator and the extension users the option of setting 'Once Only' or 'Daily' Alarms. When this flag is
disabled, the system will allow only 'Once Only' alarms to be set.
• Configurable Alarm Category - disable this flag, if you do not want the system to provide the Operator
the option of setting 'Personalized' or 'Automated' Alarm calls. When this flag is disabled, the system
will follow the 'Automated' Alarm call serving mechanism. The Operator will not be prompted to choose
between 'Automated' and 'Personalized' Alarm calls when setting Alarm calls for an extension phone.
• The flags ‘Configurable Alarm Type’ and ‘Configurable Alarm Category’ are not applicable for Voice-
guided Alarms. In the case of Voice-guided Alarms, the Operator/Extension user will be prompted to
select the Alarm type and serving mechanism, each time, even when both aforementioned flags are
disabled.
• Select an Advanced Feature Template number (by default Template 01 is assigned to all extension
phones).
• Scroll with the horizontal bar to reach the column 'Alarm Notification Type'.
• Select the desired Alarm Notification Type to be set on all extension phones: Voice Message, Music-On-
Hold, External Music Source, Routing Group.
• If you select the Alarm Notification Type = Voice Message, ensure that you assign a voice module to
'Alarm' voice message application. Please refer topic “Voice Message Applications” for more details.
• If you select the Alarm Notification Type = Music-On-Hold, no further configuration is required.
• If you select the Alarm Notification Type = External Music Source, make sure you connect a music source
to the 'Analog Input Port' of the system. Please refer 'Technical Specifications' of the Analog Input Port for
more details.
If you selected a different Routing Group, program it and enter the number of this group in the 'Alarm
Notification Group' field of the Station Advanced Feature Template.
• If you selected 'Routing Group', and have connected a device capable of playing messages when serving
an alarm call, read Customized Alarm Messaging Devices for further instructions.
• Click Submit at the bottom of the page to save the change in the Template.
• Apply the Template now configured with the Alarm Notification Type to the extension phones.
Refer the topic “Station Advanced Feature Template” for instructions on applying this template to SLTs,
DKPs and ISDN Terminals.
If you want to set different Alarm Notification Types for different stations112, it is recommended that you
program separate Station Advanced Feature Templates for each Alarm Notification Type. On each station,
apply the Template with the relevant Alarm Notification Type that you want program for that station.
112. For example, play Music-on-hold on a few extensions, pre-recorded voice messages on some extensions, music/message from an
external source on some extensions, and customized alarm greetings from external devices on others.
• In the field 'Number String', enter the strings to be replaced with on receiving the strings from the SLT.
• In the field 'Access Codes', enter the strings sent by the SLT on pressing the special function key for
'Alarms'. For example, if the SLT sends a string '51' to the ETERNITY, program the string '*161' (feature
access code for Alarms) in the field 'Number String', and enter the string '51' in the corresponding field viz.
'Access code'.
For example: To program Routing Group as Notification Type in Station Advanced Feature Template
02: Dial 5602-1-02 -12-4
To apply the Station Advanced Feature Template now programmed with the Alarm Notification Type to
extension phones, refer the topic “Customizing Station Advanced Feature Template using a
Telephone”.
• Dial command 3115-1-Macro Index to clear the Access code for the macro.
• Exit SE mode.
If the User wants to connect customized alarm messaging devices to the ETERNITY, the SE should configure the
system as instructed below:
• Connect the devices for customized alarm greetings to SLT ports only.
• Select 'SLT' as the 'Member Type' and enter the SLT 'Port Number' where the device is connected. It is
possible to configure 32 members in a single routing group.
• Open the 'Station Advanced Feature Template' page. By default Template Number 01 is applied to all
stations. The template has 'Voice Message' as default notification type. It is recommended that you
program another Template.
• Select 'Routing Group' as the Alarm Notification Type in the template you have selected for configuration.
• Enter the number of the 'Alarm Notification Routing Group' (default group: 31) in which you have
programmed the device (SLT port).
• Apply the Advanced Feature Template now configured with 'Routing Group' as Alarm Notification Type
and the number of the Alarm Notification Routing Group to the stations.
• Refer the section “Station Advanced Feature Template” for instructions on applying this template to
extension phones.
The first two parameters can be programmed using Jeeves and a Telephone.
The last two parameters, i.e. the Scheduled Alarm Report flag and the Time for the report must be set from the
SA mode.
• Go to 'Hotel Setting'.
• Go to 'Destination Port of Hotel Reports' and select the Communication Port/Printer port to be assigned.
• Click 'Submit' to save your settings. If a 'Communication Port' is selected, program the parameters of the
port.
114. Speed/Baud Rate, Data Bits, Parity, Flow Control, DSR sensing.
• Exit SE mode.
To program parameters of the COM Port assigned as the Destination Port using SE commands, refer the
topic “Communication Ports” for instructions.
To cancel Alarms,
Using Commands:
To cancel Alarms,
To set Alarm,
To cancel Alarms,
Dialing Commands:
To set Alarm,
To cancel Alarms,
• Extension users can set only automated alarms from their phones. For personalized alarms, they must
request the Operator.
• If there are multiple alarms set, alarms cannot be canceled selectively. Only the Operator can cancel
alarms selectively from SA mode.
• Alarms set on an extension will be served, even if DND is also set on the same extension.
The Operator may dial the following SA commands using EON or an SLT. It is assumed that the Operator is in
SA mode.
• Dial 1072-037.
• Dial Time in Hours and Minutes (HH:MM)
• You get a confirmatory text message and a confirmation tone.
• Go Idle or you get dial tone after 3 seconds
The system will print the Scheduled Alarm Report at the time set by the Operator at the designated
Destination port.
The SLT from which the Operator dials these commands must have the features 'Allow SA Commands' and
'System Administrator (SA) Mode' enabled in its Class of Service.
OR
• Dial 1072-913.
• You get a confirmatory text message and a confirmation tone.
• Go Idle or you get dial tone after 3 seconds
• The Alarm Report will be printed on the destination port (Communication/Printer Port) assigned.
• Open Jeeves.
• You can cancel any of the unserved Alarm calls by selecting the check-box and clicking the 'Cancel
Selected Alarms' button on this page.
• You can also print this page by clicking the 'Print' button on this page.
What's this?
Alternate Number Dialing allows you to dial different phone numbers in an attempt to reach a person whose line is
busy.
Alternate Number Dialing is useful when the person or organization you are trying to reach has more than one
number, where they may be reached. The system dials out different phone numbers of the same party, saving you
time and effort of dialing each of these numbers manually.
How it works
This feature works as an extension of the features “Last Number Redial” and “Auto Redial”. It requires you to
program Alternate Number Groups in the Global Directory first. With the alternate numbers programmed in the
Global Directory, all you need to do is to use Last Number Redial or Auto Redial, every time you want the system to
try Alternate Number Dialing.
For example: Midas Business Solutions has four telephone numbers: 2640459, 2631235, 2635589 and 2565590.
To be able to use Alternate Number Dialing, you must first program all four numbers as Alternate Number Group in
the Global Directory.
Now, when you dial one of these numbers, '2640459', and get a busy tone, you can either initiate Last Number
Redial or set an Auto Redial request.
• The system will dial an alternative number for the dialed number.
• If the redialed number is busy, you can set Last Number Redial again.
• If the second alternative number is also busy, you can set Last Number Redial again.
• This process will be repeated each time you set Last Number Redial, until the call gets through.
• If the alternative number is busy, the system will redial another alternative number.
• The system will dial a different (alternative) number on each auto redial attempt115, until the call gets
through.
115. The number of auto redial attempts depends on the Auto Redial Count programmed in the system. By default, the system will
make 5 redial attempts if Auto Redial 'normal' is set. If Auto Redial 'Priority' is set, the system will make 20 redial attempts.
• when any of the alternate numbers gets through, the system will give a ring on your extension.
(Busy) 2630555
( Bu
Calling Party sy)
2630556
2630557
ETERNITY
• Alternate Number Dialing will work only on extensions that are allowed the features “Last Number
Redial” in their “Class of Service (COS)”
• Also, Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. The Global Directory is divided into three parts, 100-399 (Part 1), 400-699
(Part 2), and 700-999 (Part 3). If an extension is assigned only Global Directory Part 2, Alternate
Number Dialing will work only for those numbers grouped as Alternate Number Groups in Global
Directory Part 2.
• Alternate Number Dialing will work also with “Abbreviated Dialing”. For example, an extension user
dials the abbreviated code 8100, and the dialed out number is busy. When the extension user sets
Redial or Auto Redial, the ETERNITY will try the alternate numbers related to 8100.
How to configure
For Alternate Number Dialing to work, the System Engineer must:
4. Enable the features 'Last Number Redial', 'Global Directory', in the Class of Service (CoS) group of the
extensions to which Alternate Number Dialing facility is to be provided. If desired, 'Auto Redial', 'Auto
Redial Priority' may also be enabled in the CoS of these extensions.
All of the above parameters can be programmed using Jeeves or dialing SE Commands from a telephone.
• To create Alternate Number Groups, the alternate numbers must exist in the Global Directory. If any of
the alternate numbers do not exist in the Global Directory, first program the numbers in the directory,
before you begin creating Alternate Number Groups. Refer the topic “Abbreviated Dialing” for
instructions on programming the Global Directory.
• Write the name of the contact on one column and the Alternate Numbers for the contact on the other
column.
• Make a list of the numbers which need to be grouped as alternate numbers. For example:
• Taking the above example further, the Alternate Number Groups on the list may be numbered as follows:
To create Alternate Number Groups and program them in the Global Directory,
• Enter the number of the Alternate Number Group in the last column of the page.
For example, you have assigned Alternate Number Group '001' to all the numbers of the contact Midas
Business Solutions, enter this number against each number belonging to this contact.
Similarly, enter Alternate Group number '004' against the numbers belonging to the 'GoodLife Inn' to which
it is assigned.
: : : : :
The numbers of the contacts may not necessarily appear alphabetically or in a sequence. It is possible that
the numbers of the same contact may be programmed at different memory locations in the Global
Directory.
In the above example, one number of the GoodLife Inn is programmed at memory location Index 104 and
the other on Index 129. Since these two numbers are grouped and assigned the number alternate group
number '004', this number must be entered against the GoodLife Inn numbers at the respective memory
location Index.
• After assigning Alternate Number Groups, click 'Submit' at the bottom of the page to save changes.
• Enable the features 'Last Number Redial' and 'Global Directory', in the Class of Service (CoS) group of the
extensions to which Alternate Number Dialing facility is to be provided. If desired, 'Auto Redial', 'Auto
Redial Priority' may also be enabled in the CoS of these extensions.
However, the default CoS Group 01 has only Global Directory Part 1 enabled.
Recall that Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. So, the Global Directory Part containing the Alternate Number Groups must
be allowed to the extensions in their Class of Service. For example, if Alternate Number Groups are
programmed in Global Directory Part 2, extensions must have Global Directory Part 2 in their Class of
Service.
If all extensions are to be allowed the Alternate Number Dialing facility, simply enable the Global
Directories containing Alternate Number groups in the default CoS group 01.
However, if Alternate Number Dialing is to be allowed to select extensions only, define a new CoS group
and prepare a new Station Basic Feature Template with this CoS group and apply it to the desired
extensions.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions.
The Station Basic Feature Template 01 does not have the features Auto Redial and Auto Redial Priority in
the default CoS group 01. If these features are also to be allowed to the extensions, enable them in the
CoS you prepare.
For example: To assign the numbers of the GoodLife Inn to an Alternate Number Group, dial:
1804-1-104-004
The number '2788856' stored at Index 104 will be assigned to Alternate Number
Group 004, which is the number assigned to GoodLife Inn.
The number '2788896' stored at Index 129 will be assigned to Alternate Number Group 004.
If you have a continuous sequence of numbers that need to be programmed in the same group, you may
dial:
• 1804-2-Memory Location Code-Memory Location Code-Alternate Group Number
Here you enter a sequence of Memory Location Codes, from 100 to 999, and the number of the
Alternate Number Group.
For example: To assign the numbers of Midas Biz to an Alternate Number Group, dial: 1804-2-100-
103-001
For example: To clear the Alternate Number Group assigned to a number of GoodLife Inn, dial: 1804-1-
104-000
The Alternate Number Group assigned to the GoodLife Inn number '2788856' stored at Index 104 will
be cleared.
For example: To clear the Alternate Number Group assigned to the sequence of numbers of Midas Biz,
dial: 1804-2-100-103-000
The Alternate Number Group assigned to the Midas Biz numbers stored in a continuous sequence
starting from Index 100 to 103 will be cleared.
• Exit SE mode.
How to use
Confirm with your System Engineer that
• Alternate Number Groups are programmed in the Global Directory allowed to your extension.
• 'Basic Features' (these include Redial) are enabled in the Class of Service allowed to your extension.
Now, follow the instructions for using the feature “Last Number Redial”.
What's this?
Auto Answer allows incoming calls to be answered without any manual interventions by the extension users.
This feature is particularly useful for Operators in high call traffic settings, as it saves them the effort of picking up
the handset or pressing the speaker key repeatedly.
How it works
With Auto Answer set on an extension DKP, whenever a call lands on the DKP extension,
• the extension rings for the duration of the Auto Answer Timer116. This timer is programmable, and by
default it is set to 1 second.
• on the expiry of the Auto Answer Timer117, the system plays a beep to the user.
• the DKP goes OFF-Hook to answer the call, without any intervention by the extension user such as picking
up the handset or pressing the speaker or the headset key.
• If a headset is connected, and headset connectivity is enabled on the DKP, the incoming speech audio will
be diverted to the headset automatically.
Auto Answer works only if the DKP is in idle state; the phone must not be busy with an active call or using a feature.
How to configure
For Auto Answer to work, you are required to do the following:
2. Change Auto Answer Timer, if required. The range of this timer is 1 to 9 seconds. By default, the Auto
Answer Timer is set to 1 second.
3. Enable Headset Connectivity flag in the DKP Parameters, if headset is to be used for Auto Answer.
All of the above can be programmed by the System Engineer using Jeeves and a Telephone.
The DKP extension users can also program the above parameters using the Phone Menu of EON. See "How
to use" Auto Answer later in this topic.
116. This timer defines the time in seconds that the DKP should wait before going OFF-Hook to answer incoming calls.
117. This timer defines the time in seconds that the DKP should wait before going OFF-Hook to answer incoming calls.
• If the DKPs are already installed and configured, identify the DKP you wish to provide Auto Answer feature
by its Hardware Port/Slot Number, Access Code or Name.
To cancel Auto Answer, disable the flag by selecting the check box.
• Now, go to the column 'Auto Answer Timer', and set the Timer as required. By default the Timer is set
to 1 second.
• If Headset is to be used by the DKP, go to the column 'Headset Connected?' and click the check box to
enable the flag.
• Repeat the above steps to program Auto Answer parameters for each DKP that is to be provided this
feature.
• Exit SE mode.
How to use
Extension users can set/cancel Auto Answer and enable Headset connectivity from their DKP by navigating the
Menu of EON.
OR
118. This function must have been programmed by the System Engineer on a DSS Key of EON. Refer "Digital Key Phone - Keys Pro-
gramming" for instructions.
OR
119. This function must have been programmed by the System Engineer on a DSS Key of EON. Refer "Digital Key Phone - Keys Pro-
gramming" for instructions.
What's this?
If the extension number you have dialed is busy or is not responding, you may use the Auto Call Back feature,
instead of repeatedly dialing the number. Similarly, when you dial a code to access a trunk and the trunk is busy,
you may set Auto Call Back.
How it works
When you set Auto Call Back,
• As soon as both extensions, yours and the remote extension, are available, the system will ring first on
your extension for the duration of the Auto Call Back Ring Timer. This timer is set by default to 30 seconds
and is programmable.
• When you go OFF-Hook, the system will ring on the remote extension (provided it is also available at that
moment) for the duration of the Auto Call Back Ring Timer.
• When the remote extension user goes OFF-Hook, your call will get connected.
However, if the remote extension gets busy before the system can ring on it, the system will continue to try
again.
Auto Call Back set for a busy trunk works the same way. As soon as the busy trunk port you are trying to
access is available, the system will ring your extension. When you go OFF-Hook you will be connected to
the trunk port.
• Each extension of the ETERNITY can set only one Auto Call Back request at a time. If you set another
Auto Call Back request, before the first one has been served, the system will override the first request
and serve the second.
• The ETERNITY has the capacity to serve 300 Auto Call Back requests from its extensions at a time.
The service duration for each request is 60 minutes. Requests that are not served within 60 minutes
are automatically cancelled by the system. Also, the system will not serve any more requests if all the
300 requests are pending. In such a case, the system will play an error tone, when an extension
attempts to make a request.
Auto Call Back request set by you will be cleared by the system if:
• it was successfully served, i.e. your extension was connected to the remote extension or the trunk you
were trying to reach.
• you do not answer the Auto Call Back ring, before the expiry of the Ring Timer, i.e. within 30 seconds
(default setting).
• the remote extension does not answer the Auto Call Back ring before the expiry of the Ring Timer.
• Auto Call Back works for internal calls and for accessing trunk ports only.
• Internal calls include calls between PBXs that are networked using Q-SIG.
How to configure
Auto Call Back is a Class-of-Service dependant feature. An extension user can set/cancel Auto Call Back only if it
is enabled in the extension's Class of Service.
The only programming involved in this feature is enabling/disabling Auto Call Back in the Class of Service and
changing the duration of the Auto Call Back Ring Timer, if required.
However, if Auto Call Back Busy/No Reply is to be denied to any of the extensions, follow these steps:
1. Define a CoS group with Auto Call Back Busy/No Reply disable.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the selected extensions to which Auto Call Back is to be denied.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions on how to
enable/disable a feature in a CoS group, how to prepare a Station Basic Feature Template with a new CoS
group and assign the new template to SLT, DKP and ISDN Terminal extensions using Jeeves.
If the User wants to increase or decrease the duration of the of the Auto Call Back ring on both extensions, i.e.
the extension requesting Auto Call Back and the destination extension, program the 'Auto Call Back Ring
Timer', according to User preference.
To program Auto Call Back when Busy/No Reply in a CoS group, dial:
• 1302-1-COS Group-Feature Number-Code
Where,
CoS group is from 01 to 20
Feature Number for Auto Call Back when Busy is '04'
Feature Number for Auto Call Back when No Reply is '05'.
Code is
0 for Disable
1 for Enable.
For example: To enable Auto Call Back when Busy in CoS group 02, dial 1302-1-02-04-1
To enable Auto Call Back when No Reply in CoS group 02, dial 1302-1-02-05-1
For example: To apply CoS group 02 with Auto Call Back Busy and No Reply for each Time zone in
Station Basic Feature Template 02, dial the following commands:
5502-1-02-03-02 for Working hours
5502-1-02-04-02 for Break hours
5502-1-02-05-02 for Non-working hours
• To apply the Station Basic Feature Template now programmed with Auto Call Back when Busy and No
Reply, to the extensions, dial the following commands:
• Exit SE mode.
How to use
Extension users can set two types of Auto Call Back:
• Auto Call Back on Busy - when the extension/trunk they are trying is Busy.
• Auto Call Back on No Reply - when there is no reply from the extension they are trying.
Auto Call Back can be set from EON as well as any SLT.
• Press the 'Call Back' Key (on EON48) or the 'Auto Call Back' Key (on EON42) on Busy Tone.
• You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key will
be turned on.
• Go idle or you get dial tone after 3 seconds.
• Press the 'Call Back' Key (on EON48) or the 'Auto Call Back' Key (on EON42) again.
• You get confirmatory message "Auto Call Back Canceled" on the phone's display. The LED of the DSS
Key will be turned off.
• Go idle or you get dial tone after 3 seconds.
Using Command:
• Dial 102.
• You get confirmatory message 'Auto Call Back Canceled' on the phone's display. The LED of the DSS key
assigned to Auto Call Back will be turned off.
• Go idle or you get dial tone after 3 seconds.
• On Busy Tone.
• Dial 2.
• You get confirmatory tone
• Replace handset or you get dial tone after 3 seconds.
• Press the 'Call Back' Key (on EON48) or the 'Auto Call Back' Key (on EON42) on Ring Back Tone.
• You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key will
be turned on.
• Go idle or you get dial tone after 3 seconds.
• Press the 'Call Back' Key (on EON48) or the 'Auto Call Back' Key (on EON42) again.
• You get confirmatory message "Auto Call Back Canceled" on the phone's display. The LED of the DSS
Key will be turned off.
• Go idle or you get dial tone after 3 seconds.
Using Command:
• Dial 102.
• You get confirmatory message 'Auto Call Back Canceled' on the phone's display. The LED of the DSS key
assigned to Auto Call Back will be turned off.
• Go idle or you get dial tone after 3 seconds.
If you hear an error tone while setting an Auto Call Back request, it is likely that the system already has 300
pending requests and is unable to accept yours.
What's this?
The Auto Redial feature retries a call automatically if the dialed number is busy. It repeatedly checks the busy line
till it is free. When the called number is no longer busy, the extension of the caller rings.
Auto Redial saves time and the effort of repeatedly dialing the entire phone number over and over until the called
party gets off the phone.
How it works
When an extension user dials a number and gets a busy tone, s/he may set Auto Redial. When Auto Redial is set,
• It waits for the 'Dial Tone Wait Timer120' to expire to begin sensing the dial tone from the PSTN/CO
Network. This timer is programmable, and is set to 3 seconds as default.
• On sensing the dial tone the ETERNITY will dial out the requested number and will wait until the 'Ring
Back Tone Wait Timer121' expires to sense the Ring Back Tone from the requested number. This timer is
programmable and is set to 60 seconds as default.
• If the system does not detect Ring Back Tone for 60 seconds, it releases the trunk and tries again after
some time. If the system detects a busy tone, it releases the trunk and redials the number automatically
after some time. This process is repeated until the system detects the Ring Back Tone.
• When the ETERNITY detects the Ring Back Tone instead of the Busy Tone, it will ring on the extension
that set Auto Redial. The extension will ring for the duration of the 'Redial Ring Timer122'. This timer is
programmable and is set to 45 seconds as default.
• If the extension is in the middle of any activity such as dialing, ringing or speech, the ETERNITY will
suspend Auto Redial until the extension becomes idle again. After which it dials the requested number
again.
Two types of Auto Redial are supported by the ETERNITY - Auto Redial (normal) and Auto Redial 'Priority' - that
differ from each other in terms of the number of redial attempts and the interval between attempts.
• Auto Redial (normal): The system is programmed by default to make 5 attempts to redial at an interval of
45 seconds (default) between each attempt. Both, the number of attempts as well as the duration of the
120. Time for which ETERNITY waits to sense the dial tone from the PSTN/CO Network. Valid range of the timer: 000 to 255 seconds.
Default: 003 seconds.
121. Time for which ETERNITY waits to sense the RBT from the PSTN/CO Network after dialing the requested number. This timer is
particularly relevant to TWT ports. Valid range of the timer: 000 to 255 seconds. Default: 060 seconds.
122. Time for which the extension that has requested Auto Redial should ring. Valid range of the timer: 000 to 255 seconds. Default:
045 seconds.
• Auto Redial 'Priority': the system makes a greater number of attempts to redial and the duration of the
interval between each attempt is less. By default the system is programmed to make 20 redial attempts at
intervals of 20 seconds. The number of attempts as well as duration of the interval are programmable; for
instance, the number of attempts can be set to 30 and the interval to 15 seconds.
To change the number of redial attempts and the interval between them, the SE must Auto Redial Count and the
Auto Redial Timer respectively. In addition to these, the system has three other related timers, which can be
programmed to match User preference:
• An extension user can request Auto Redial for multiple numbers at a time from the same extension and
more than one extension can attempt auto redial simultaneously.
• The system uses the same OG Trunk Bundle Group you used. If you dialed the number on group code
60, the system grabs one of the free trunks from group code 60 for Auto Redial.
• If the number was dialed the first time using selective trunk access, the system will use the same trunk
to execute Auto Redial.
• If the extension is programmed for 'Dynamic Lock', and you have set the 'Auto Redial', the system will
check the Toll control as per dynamic lock level.
Auto Redial may not work well on Two-wire Trunk lines, as its functioning greatly depends on line
condition. Unlike ISDN, GSM and VoIP trunks, the line condition of TWT trunks may not always measure
up to the standard requirement for Auto Redial to function.
How to configure
For Auto Redial to work, the System Engineer must:
1. Enable the features 'Auto Redial' and 'Auto Redial Priority' in the Class of Service (CoS) group of the
extensions to which this feature is to be allowed.
2. Change the 'Auto Redial Normal/Priority Count' and the 'Auto Redial Normal/Priority Timer' to match User
preference. This will change the number of redial attempts made by the system and the interval between
them.
3. If required, also change other related Timers such as Auto Redial Dial Tone Wait Timer, Auto Redial Ring
Back Tone (RBT) Wait Timer, Auto Redial Ring Timer.
All the above parameters can be programmed using Jeeves and by dialing SE commands from a
telephone.
If the User wants to allow all extensions the Auto Redial and or the Auto Redial Priority feature, the you can simply
enable this feature in the default CoS group 01.
However, if Auto Redial/Auto Redial Priority is to be allowed on only select extensions, follow these steps:
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the extensions to which Auto Redial/Auto Redial Priority is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions.
• You may change any of the related timers - Auto Redial Dial Tone Wait Timer, Auto Redial Ring Back Tone
(RBT) Wait Timer, Auto Redial Ring Timer - as per your preferences on this page.
i. The Timer for Auto Redial Normal as well as Priority must be set to more than 5 seconds.
ii. The Auto Redial Priority Count should be set to less than 15.
• enable a feature (in this case Auto Redial and Auto Redial Priority) in the CoS group.
• apply the Template with the on DKP and SLT extensions, using SE commands.
• Exit SE mode.
How to use
Auto Redial can be set/canceled from EON as well as SLT.
Using Command:
What's this?
You can connect and operate a variety of gadgets such as a door lock, a siren/hooter, a school bell, a water pump,
sprinklers, automated illuminations (office lights, porch and terrace lights, glow signboards, street lights), and
others as automated control applications.
The “Digital Output Port (DOP)” of ETERNITY is used for running the automated control applications. Any external
device that meets the technical specifications of the DOP can be connected to it.
The gadget connected to the DOP can be operated, i.e. switched ON and OFF, manually by dialing Feature
Commands, or automatically at predefined time periods or on instigation from the “Digital Input Port (DIP)”.
The Automated Control Application gadget can be operated also from a remote location using “Direct Inward
System Access (DISA)”.
The ETERNITY PE supports 3 DOPs (located on the Door Phone Card), allowing you use 3 different automated
control application gadgets.
ETERNITY ME and GE support a single DOP each. So, you can connect and operate only a single application on
these models and their variants.
How it works
The gadget connected to the DOP can be operated in 9 different ways, referred to as Gadget Operation Modes.
Each of these is described in the table below:
Gadget
Description Typical Application
Mode No.
1 Switch ON and Switch OFF the gadget using Command. Office Lights.
The gadget connected to the DOP can be operated any Lights can be switched on arrival of the
time as required, by dialing a command to switch ON. staff, and switched off when all staff
have left.
The gadget is switched OFF by dialing a command.
3 Switch ON and Switch OFF the gadget as per the Flip- Festive illuminations.
Flop Interval.
Strings of light bulbs can be turned on
The gadget remains ON for the duration of the Timer and turned off in the intervals of a few
called Flip-Flop Interval and remains switched OFF for seconds to get a blinking effect.
the duration of the same Timer.
4 Switch ON the gadget using command and Switch OFF Glow signboard.
at the Preset OFF time.
Glow signboards can be turned ON
The gadget is turned ON by dialing a command. It is manually when required. Regardless of
turned OFF at a specific time called the Preset OFF- when they were turned on, they can be
Time. turned OFF automatically at a particular
time (for example, 7am).
The Preset OFF-Time is programmable. It is to be
programmed in the 24 Hours, Minutes and Seconds For example, you can turn on the Glow
format (HH:MM:SS) signboards in the evenings whenever
you remember to and have them
switched OFF early next morning.
6 Switch ON and Switch OFF the Gadget as per the A school bell or a factory siren/bell.
Schedule.
A school bell must ring multiple times
The gadget is switched ON and OFF according to the during a day at regular intervals to
start and end time programmed. indicate break time and end of class.
Similarly, a factory siren/bell must ring
The Scheduled ON and OFF Time are to be at regular intervals during a day to
programmed in the 24 Hours, Minutes and Seconds indicate shift change and break time.
format (HH:MM:SS).
The bell/siren connected to the DOP
ON and OFF Time schedules can be programmed for 24 can be programmed to start and stop
hours in a day. ringing at defined intervals during the
day.
7 Switch ON the gadget following instigation from DIP and A Water Pump.
Switch it OFF at next instigation from DIP.
When the water tank goes empty, the
water level sensor connected to the DIP
senses it and sends an instigation to
switch ON the Pump connected to the
DOP.
9 Switch ON the gadget following instigation from DIP and Porch, Terrace, and other Lights on or
Switch OFF after a Preset Interval. around the premises.
The Preset Interval is programmable. By default it is set An object sensor connected to the DIP
to 10 seconds. senses the presence of persons/
objects. The DIP sends instigation to
the DOP, which turns on the Lights.
Similarly, when the Gadget mode is Switch ON and OFF on Instigation from DIP, you can still dial the
Feature Command to switch ON. The system will turn ON the gadget without waiting for instigation
from the DIP. The system will wait for instigation from DIP to turn OFF the gadget. But if you dial the
feature Command to switch OFF, the system will turn OFF the gadget without waiting for instigation
from DIP.
• ETERNITY supports only one Schedule for all gadgets. If you want to operate more than one gadget
(possible only on ETERNITY PE, as it supports 3 DOPs) as per the "Schedule" mode, the Schedule
you program will be applied commonly to all the gadgets.
• When the DOP is programmed to be switched ON/OFF on instigation from the DIP, the trigger time of
the DIP will be the same as the Event Sense Timer of the DIP. Refer “Digital Input Port (DIP)” to know
more about this Timer.
• Automated Control Applications operated in the 'Preset ON-Time', 'Preset OFF-Time' and 'Scheduled'
modes function on the basis of the “Real Time Clock (RTC)” of the ETERNITY. Though the RTC circuit
automatically updates the date, day and time values, it may drift over a long period. Check the RTC
values every month and reset the values to correct the drift.
If you want to use an application that requires an input sensor device, connect the sensor device to the Digital Input
Port.
For instructions refer the topics “Installing ETERNITY ME”/“Installing ETERNITY GE”/“Installing ETERNITY PE” as
relevant to your model of ETERNITY. Also refer the topic “Digital Output Port (DOP)” and “Digital Input Port (DIP)”.
How to configure
Programming of automated control applications involves the following:
• Configuring the Digital Output Port to which the gadget is connected. Refer the topic “Digital Output Port
(DOP)” for instructions.
• Configuring the Digital Input Port, if you want to operate the gadget in conjunction with a device connected
to the DIP. Refer the topic “Digital Input Port (DIP)”.
• Configuring the parameters of the gadgets attached to the DOP of ETERNITY.
• Gadget mode: Select gadget mode in which you want to operate the device. Refer the description of
"Gadget Operation Modes" earlier in this topic.
• DOP Number: Select the DOP number to which the gadget is connected.
• Preset Interval: You may program this Timer if you have selected the option "Switch ON the gadget
using Command and Switch OFF after a Preset interval" or the option "Switch ON the gadget following
instigation from DIP and Switch OFF after a Preset Interval" as Gadget mode.
The range of this timer is from 001 to 255 seconds. By default it is set to 10 seconds.
• Flip-Flop Interval: You may program this Timer if you have selected the option 'Switch ON and Switch
OFF the gadget as per the Flip-Flop Interval' as Gadget mode.
• Preset ON-Time (HH:MM:SS): You may program this Timer, if you have selected the option 'Switch
ON the gadget at Preset ON time and Switch OFF using command' as Gadget mode. The time format
is in 24 Hours: Minutes: Seconds format. Select the hours, minutes and seconds in the respective
boxes.
• Preset OFF-Time (HH:MM:SS): You may program this Timer, if you have selected the option 'Switch
ON the gadget using command and Switch OFF at the Preset OFF time' as Gadget mode. The time
format is in 24 Hours: Minutes: Seconds format. Select the hours, minutes and seconds from the
respective boxes.
You can program different ON and OFF Time schedules for 24 hours in a day. So you can run the
control application more than once in a day at different times.
Each Time Schedule - ON and corresponding OFF Time - is stored against an Index number from 1 to
24. The system stores each ON and OFF Time Schedule in ascending order of time. If you want to run
the control application more than once in a day, program the Time Schedule in the sequence from
earliest to last against each Index, i.e. morning hours should be programmed before evening hours
against each Index number.
• Click 'Submit' at the bottom of the page to save your DOP settings
• Repeat the same steps to program the parameters of Gadget-2 and Gadget-3 (applicable for
ETERNITY PE only).
123. Refer the table 'Gadget Operation Modes' at the beginning of this topic for description of the modes and their numbers.
• Exit SE mode.
Users need to dial Feature Commands if the selected Gadget mode requires it or whenever they want to override
the gadget mode selected to operate the gadget. For this, they must dial feature commands to turn ON and turn
OFF the DOP to which the gadget is connected.
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
However, you can operate gadgets from the DISA mode using only the Feature Commands.
When the control application gadget is set to operate on the basis of a Timer or a Schedule or Instigation,
you must inform all users about the duration of the Timers/Schedule, so that they can act accordingly
whenever required.
For example, if a door lock is set to close automatically after the Preset Interval of 30 seconds, the users of
this application must be aware of this, so that s/he may exit the room/building before the expiry of this
timer. Or the Door will be automatically closed/locked.
What's this?
ETERNITY offers connectivity to different networks - PSTN, GSM, ISDN T1E1PRI, BRI, VoIP - each having a
different numbering plan. For example, the GSM network requires area codes to be dialed also for local numbers,
whereas PSTN requires dialing of area codes for long distance calls.
When ETERNITY is connected with multiple networks, outgoing calls may be routed through any of these
networks, depending on the routing pattern configured in the system. However, as extension users do not know
through which telecom network their calls will be routed, they cannot be expected to dial numbers according to the
numbering plan of the destination networks.
The feature, Automatic Number Translation of ETERNITY takes care of this. It modifies/manipulates dialed
numbers or part thereof to match with specific route numbering plan understood by the destination network (PSTN,
GSM, VoIP). This includes adding or stripping of country codes, area codes.
For example, when an extension user dials a local landline number, if Automatic Number Translation is so
programmed, the ETERNITY will prefix the number with the appropriate country-area code when it routes the call
through the GSM network.
How it works
Automatic Number Translation makes use of the Automatic Number Translation Table which comprises two pre-
defined Number lists:
• Dialed Number List - contains the numbers dialed by the extension users
• Substitute Number List - contains the corresponding numbers for those on the Dialed Number List that
the system will dial out as the destination numbers.
Both lists must be programmed, and the Table must be applied to the trunk ports, through which calls are
routed to the destination networks. The trunk ports may be TWT, Mobile, VoIP, depending on system
configuration.
You can program different 8 Automatic Number Translation Tables; each table accommodates 32 Dialed
Numbers and their corresponding Substitute numbers.
For example:
• Automatic Number Translation (ANT) Table-1 has '95' programmed as Dialed Number in Index-01 and '91'
as the corresponding Substitute Number at Index-01.
How to configure
The working of the Automatic Number Translation feature is controlled by two parameters: 'Automatic Number
Translation Table' and 'Automatic Number Translation flag' in the Outgoing Trunk Bundle
Decide the number of ANT tables you need to program. You can program 8 different ANT Tables, with a maximum
of 32 number strings in each.
For convenience of programming, draw three column tables on paper. In each table, enter the Index numbers 1 to
32 in the first column, Dialed Number in the second column, and Substitute Number in the third column.
Enter the Dialed Numbers and their corresponding Substitute Numbers against each Index number. For example:
'95' is entered as the Dialed Number at Index 01 and its Substitute Number '91' also at Index 01.
Now, configure these tables in the system using Jeeves or dialing SE commands from a telephone.
• Now, select the ANT Table number you want to assign to this trunk.
For example: To configure ANT Table 1 with the number string '95' as dialed number at Index 01, dial
4751-1-1-01-95-#*
For example: To configure '91' as the substitute number for '95' at Index 01, in ANT Table 1 (see the
previous example), dial: 4751-1-1-01-91-#*
To program Special Digits, refer to the table below for the codes to be dialed:
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
.(dot) #9
# ##
* **
W *1
When programming special digits from an SLT or EON, press #9 for ‘.' (dot/period).
But for dialing numbers with '.' (dot/period) from an SLT or EON, press * (star).
For example:
To enable Automatic Number Translation flag in OGTB number 1, dial: 6702-1-001-5-1
To enable the same flag in OGTB numbers 1 to 8, dial: 6702-2-001-008-5-1
To enable the same flag in all OGTBs, dial: 6702-*-5-1
For example:
To assign ANT Table 3 to OGTB number 1, dial: 6702-1-001-6-3
To assign ANT Table 3 to OGTB numbers 1 to 8, dial: 6702-2-001-008-6-3
To assign ANT Table to all OGTBs, dial: 6702-*-6-3
• Exit SE mode.
What's this?
Extension users can be played pleasant music on their phones, with the 'Background Music' feature of the
ETERNITY.
The advantage of this feature is that extension users can listen to pleasant music on their extensions as they work,
without affecting communication (incoming and outgoing calls) on their extensions.
How it works
• Music is played after the extension user dials the Background music feature code and goes ON-Hook.
• Music is stopped automatically whenever there is an activity on the extension phone, such as:
• an incoming call landing on the extension. (Music is stopped and the phone rings).
• the extension user going OFF-Hook to make an outgoing call. (Music is stopped and the system dial
tone is played.)
• the extension user goes OFF-Hook to access any system feature using the phone.
• Volume of the background music can be controlled using the Volume keys of the DKP.
• The extension user must first press the Speaker key. The system interprets this as 'OFF-Hook' and plays
dial tone.
• The extension user must dial the Background music feature code and press the Speaker key again. The
system interprets this as 'OFF-Hook'. It plays the dial tone and waits for the First Digit Timer to elapse.
Background Music is played only after the First Digit Timer has elapsed.
• Music is stopped when there is an incoming call. The extension user is played Ring Back Tone.
• Once the call has ended, the extension user can go ON-Hook. If the extension goes OFF-Hook, Music will
be played again at the end of the dial tone and the First Digit Timer.
• However, if the extension user dials a feature access code/extension number/external number before the
end of the First Digit Timer, music will not be played, until the extension goes OFF-Hook again.
• Volume of the music can be controlled using the volume keys of the SLT.
The external device must be connected to the Analog Input Port (AIP) of the ETERNITY. The external device must
comply with the technical specifications of the AIP.
The volume must be set to a level such that the music is not very low or very loud. The volume of the signal coming
from this device must never increase beyond the specified limits of the AIP. This may result in permanent damage
to the system. Matrix Warranty does not cover damages resulting from improper use.
For installation instructions refer “Installing ETERNITY ME”, “Installing ETERNITY GE”, “Installing ETERNITY PE”,
as applicable to your model.
• Background Music can be played only on extensions that have this feature enabled in the “Class of
Service (COS)” assigned to them.
How to configure
For the Background Music feature to work, the System Engineer must:
1. Enable Background Music in the Class of Service of the extensions to which this feature is to be allowed.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of
ETERNITY. The Station Basic Feature Template 01 has the feature Background Music enabled in the default
Class of Service (COS) group 01. So, all extension users of the ETERNITY can play Background Music,
provided that their phones are a DKP or an SLT with Speaker.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the Station to which Background Music is to be denied.
How to use
Background Music can be played by extension users whose phone is a DKP or an SLT with Speaker function.
What's this?
Station Message Detail Records (SMDR), i.e. records of internal, incoming and outgoing calls made to/from
extensions of the ETERNITY are stored124 by the system in the 'SMDR Buffer'. The SMDR Incoming Call buffer
has a capacity of storing a maximum of 5000 incoming call records. The SMDR Internal Call buffer can store up to
1000 internal call records, while a maximum of 6000 outgoing call records can be stored in the SMDR Outgoing
Call buffer.
SMDR buffer data can be cleared by the SE or SA manually or the system clears the data automatically when the
SMDR buffer is full, by replacing the oldest call record with the latest (First In First Out logic).
While the SMDR buffer data is maintained even during power failures, accidental data loss is not an uncommon
occurrence.
Therefore it is advisable to Back-Up SMDR records to restore accidentally deleted, lost or corrupted files.
Back-up of SMDR records can be stored on a PC for retrieval later. The ETERNITY provides an embedded
FTP server125 to transfer SMDR call records on a PC.
You can backup-SMDR call records over the FTP using Jeeves.
The ETERNITY stores call records (SMDR) in the text format so that the files are readable when downloaded
using Jeeves.
• Enter the IP Address of the Ethernet Port of Eternity in the Address Bar.
124. ETERNITY will store SMDR only if the SMDR-Storage flag has been enabled. The call records are stored according to the Storage
filters set.
125. File Transfer Protocol (FTP), is a standard network protocol, used to exchange and manipulate files over a TCP computer network
such as the Internet. FTP is commonly used to transfer Web server for everyone on the Internet. It is also commonly used to down-
load program and other files to your computer from other servers.
• On the left navigation bar, scroll down to reach 'Station Message Detail Recording' and click this link.
• Select all files/files you would like to store. Click 'Copy selected items'.
While uploading SMDR files on to ETERNITY, first, remove the current files in the system. Copy the new
files from computer (backup source) on to the system.
You can tag the back-up folders on the PC by date to store the records as archives.
ETERNITY stores the current settings of hardware and software features in the System Configuration data files.
Therefore it is advisable to Back-Up System Configuration files to restore original system configuration.
System Configuration files can be stored on a PC. The ETERNITY provides an embedded FTP server126 to
transfer Configuration files on to a PC.
System Configuration files can be transferred to a PC over the FTP using Jeeves.
• Enter the IP Address of the Ethernet Port of ETERNITY in the Address Bar.
• On the left navigation bar, scroll down to reach 'Configuration Upload'. Now, click this link.
126. File Transfer Protocol (FTP), is a standard network protocol, used to exchange and manipulate files over a TCP computer network
such as the Internet. FTP is commonly used to transfer Web server for everyone on the Internet. It is also commonly used to down-
load program and other files to your computer from other servers.
• On successful login the FTP window will open. The all the system configuration files will appear with the
extension '.cfg'.
• Select the path where you want to store the back-up files on the PC. Click 'Copy' button.
• Log out after the Back-up is completed or log in again to use the web pages for further programming.
You can archive the Back-up of configuration files by tagging the back-up folders on the PC by date.
The ETERNITY System Software may be accidentally deleted or corrupted during maintenance and upgrade
procedures. To prevent data loss, it is advised to back-up System Software.
This can be done by storing the System Software files on a PC, using the embedded FTP server provided by the
ETERNITY.
System Software files can be transferred to a PC over the FTP using Jeeves.
• Enter the IP Address of the Ethernet Port of Eternity in the Address Bar.
• On successful login the FTP window will open. The all the system software files will appear on the window.
• Select all files/files you would like to store and click 'Copy selected items'.
• Select the path where you want to store the back-up files on the PC. Click the 'Copy' button.
• Log out of Jeeves after the Back-up is completed or log in again to use the web pages for further
programming.
What's this?
Barge-In allows you to break into an on-going conversation between two extension users, between an extension
user and an external caller as well.
Barge-In can be used by Operators to transfer Incoming calls to busy extensions. The Operator can put the caller
on hold, barge into the busy extension to inform about the call, and then transfer the call.
ETERNITY offers flexibility to allow/deny Barge-In feature to an extension user, i.e. allow the extension user to
barge into on-going conversations. It also provides the flexibility to prevent conversations of extension users from
being barged in, referred to as Privacy against Barge-In.
How it works
• A, B and C are users of the system.
• C calls A.
• C gets Ring Back tone (RBT) and A gets beeps indicating a new call. If A is using EON, C's name and
number appear on C’s phone display.
• C gets RBT and A gets beeps for Barge-in timer. (By default, 10 seconds)
• If A does not respond till the end of the Barge-In Timer (set to 10 seconds, by default), A gets connected to
C. B is put on hold and is given hold-on music.
• If B disconnects while A and C are talking, the held call between A and B is cleared.
• If B keeps holding the call and C disconnects, the call between A and C is cleared and A is connected back
to B.
• If B keeps holding the call and A disconnects, the call between A and C is cleared and A gets ring. A picks-
up the handset and gets connected back to B.
Feature Interactions
• Call States:
• Barge-In works only if the user about to be barged in is in a two-way normal speech with another user
or external party.
• It will not work if the busy signal is due to the user being OFF-Hook, or in the middle of dialing, or
accessing a feature of the PBX.
• “Call Toggle”: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
• Privacy against Barge-In: If the feature 'Privacy against Barge-in is enabled for an extension, it cannot be
barged into.
• “Priority”: No Interaction with Barge-In. If 'A' has lower priority than 'B' but has Barge-In enabled; A can
barge in B.
• “Do Not Disturb (DND)”: Barge-In will not work if the called user has set DND. If 'A' has set DND. A is
busy with C. B calls A. B cannot barge in A.
• “DND-Override”: Barge-In will work if the calling user is allowed DND-Override and also has higher
'Priority' than the called user. If 'A' has set DND. A is busy with C. B calls A. On busy signal, B dials the
Barge-In code. Barge-In will be successful only if B has DND-Override enabled and has higher priority
than A.
How to configure
The functioning of this feature is controlled by three parameters, 'Barge-In', 'Privacy against Barge-In' and 'Barge-In
Timer'.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the stations of
ETERNITY. The Station Basic Feature Template 01 is assigned CoS group 01. The default CoS group 01 has both
Barge-In and Privacy from Barge-In are disabled. So, none of the stations of the ETERNITY can use these
features.
If you want to allow Barge-In to the all extensions, simply enable Barge-In in the default CoS group 01.
However, if Barge-In is to be allowed on only select extensions then follow these steps:
b. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions on
programming.
Barge-In Timer
Barge-In Timer is the time after which the caller gets connected to the called party. By default the Timer is set to 10
seconds.
How to use
OR
• Dial 4127.
• You get Ring Back Tone.
• Wait for the system to connect you to the called extension.
• Talk.
• Replace the handset after the conversation has ended.
127. This default feature access code can be changed to suit your preference. Refer the topic “Access Codes”.
What's this?
BCCH Selection feature enables you to lock the Mobile Port of ETERNITY to a particular cell or channel or BTS
(Base Transceiver Station) for various reasons such as:
Cell Locking feature will work on SIMCOM 3G engine (Version V1.19/B11 or later and with 2G networks)
only.
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic “License Management” to know more.
How it works
In the GSM network, each BTS is assigned one particular channel called as ARFCN (Absolute Radio Frequency
Channel Number), which is transmitted by BTS in BCCH (Broadcast Control Channel).
Now, when ETERNITY is switched on, the Mobile Port gets registered with the network on a particular BTS which
has the highest signal strength. However, the signal strength is not consistent. It keeps fluctuating, resulting in call
drop or poor voice quality.
Therefore, to avoid this, ETERNITY enables you to lock the Mobile Port to a particular cell or channel manually
after checking Signal Strength and Signal Quality of each cell.
How to configure
You can lock Mobile Port to a cell or a channel only through Jeeves.
• Mobile Port Number: This is number of the Mobile port for which BCCH Selection status is displayed.
You can choose a different Mobile Port number from the drop down list. The page will display the
BCCH Selection related parameters for the selected mobile port.
• Mobile Port Name: This is the name you have assigned to the Mobile port in the Mobile port
parameters.
STATUS DESCRIPTION
GSM Displayed when GSM module is in initialization state i.e. before SIM
Initialization detection.
SIM Absent Displayed when SIM Card is not detected by the system.
Registering Displayed when the Mobile Port is in registration process with the Network.
Idle Displayed when the Mobile Port is registered with the Network and it is
free.
Busy Displayed when any active call is present on the Mobile Port.
• BCCH Locking Status: The current BCCH Locking status of the mobile port is displayed in this field.
Given below is a description of the various BCCH Locking status indication messages that will appear
in this field.
STATUS DESCRIPTION
Trying to Lock Displayed when user selects Manual BCCH Locking as 'No' from 'Yes' and
module is in initialization process after system or module restart.
Trying to lock on Displayed when BCCH Locking is selected as Manual and the Mobile Port is in
BCCH xxxxx the registration process with the Network. xxxxx is the BCCH selected by the
user for locking the cell.
Manually Locked Displayed when BCCH Locking is selected as Manual and Mobile Port is
on BCCH xxxxx successfully registered with the Network. xxxxx is the BCCH selected by the user
for locking the cell.
Auto Locked on Displayed when BCCH Locking is selected as Auto and Mobile Port is
BCCH xxxxx successfully registered with the Network. xxxxx is the BCCH of the Main Cell.
xxxxx is updated as per the changes in the Main Cell's BCCH.
• Main Cell- Bit Error Rate (%): Bit Error Rate of the Main Cell is displayed in this field. Bit Error Rate
(BER) is the percentage of received bits on a digital link that are in error relative to the number of bits
received. Bit Error Rate is calculated from the received signal quality.
• Manual BCCH Locking: This parameter allows you to lock the Mobile Port to a particular cell of your
preference. By default, manual BCCH locking is set to 'No'. When manual BCCH locking is set to 'No',
Mobile Port gets locked to the cell as per the highest signal strength. Select 'Yes' if you want to lock the
Mobile Port to the particular cell selected by you.
• Auto Refresh: Click this button to refresh BCCH Selection page. All parameters on this page will be
downloaded automatically after every 15 seconds. By default, Auto Refresh button is enabled.
• Cells: Indicates the cells with which the Mobile Port can be locked. You can decide to lock the Mobile Port
with a particular cell after considering the following cell related parameters, which appear on the page:
• MCC-MNC: In this field, MCC-MNC of a cell is displayed. Mobile Country Code (MCC) is a three digit
number uniquely identifying a country and Mobile Network Code (MNC) is either a two or three digit
number used to identify a given network from within a specific country.
• LAC (Location Area Code): In this field, LAC (Location Area Code) is displayed. LAC uniquely
identifies a location area within a GSM PLMN (Public Land Mobile Network). The maximum length of
LAC is 16 bits ranging from 0 to 65535. LAC is displayed in hexadecimal characters for SIMCOM-2G
and Wavecom-2G engines which ranges from 0000 to FFFF. For SIMCOM-3G engine, LAC is
displayed in decimal digits which ranges from 00000 to 65535.
• Cell ID: In this field, Cell ID is displayed. It is a 16-bit identifier that identifies the cell. Cell ID is
displayed in hexadecimal characters for SIMCOM-2G and Wavecom-2G engines which ranges from
0000 to FFFF. For SIMCOM-3G engine, Cell ID is displayed in decimal digits which ranges from 00000
to 65535.
• BSIC (Base Station Identification Code): In this field, BSIC (Base Station Identification Code) is
displayed. BSIC allows a mobile station to distinguish between different neighboring base stations.
BSIC is a three-digit value ranging from 0 to 255.
• BCCH (Broadcast Control Channel): In this field, the BCCH value of the cell is displayed. BCCH
defines the frequency channel number.
• Receive Level: In this field, the Receive Signal Strength level of the cell is displayed. It is the average
Receive Signal Strength of the cell. Its value ranges from -110 dBm to -47 dBm.
• Manual Cell Locking: This radio button is for locking a Mobile Port to a selected cell manually.
• Select the desired Mobile Port Number from the drop down list.
• Set the parameter 'Manual BCCH Locking' to 'Yes'.
• Go to the Cell to which you want to lock the Mobile Port you selected.
• Select the radio button 'Manual Cell Locking' of that Cell.
• Click 'Submit' at the bottom of the page.
• The BCCH Locking for the selected Mobile Port will appear on this page, if Auto Refresh is enabled.
• If you have stopped Auto Refresh, click 'Refresh' at the bottom of the page to refresh the page and view
the current BCCH Locking settings of the selected Mobile port.
• You may now log out of Jeeves.
Example:
Consider the following example when using this feature:
Problem:
• ETERNITY is installed in roaming area, where more than one network is available, say A and B.
• Mobile Network Selection is set to 'Manual' mode and the first priority is programmed as network A and the
second priority is programmed as network B.
Solution:
• In this situation, user should set Manual BCCH locking mode to 'No' to register Mobile Port with the
suitable network automatically.
• Later, the user can set the Manual BCCH locking mode to 'Yes' and lock the Mobile Port to the desired cell
after assessing the cell information.
What's this?
It is common for small and medium PBXs to be connected to larger PBX systems, where the trunks of the larger
PBX are connected to the stations of the smaller system. This is usually done for the purpose of expanding the
capacity of the large PBX already in use.
How it works
Consider the following illustration.
21 S1 T1
T2
22 S2
T3
23 S3 T4
T5 S4 S1 31
T6 S5
S2 32
T7 S6 PSTN
S3 33
PBX A
Sn Tn S7
ETERNITY T1 S8
T2 S9
PBX-A is connected behind ETERNITY. In this 'Behind the PBX' configuration, the Trunk Lines T5, T6, T7 of
ETERNITY are connected to the Stations (SLT) S4, S5, S6 of PBX-A.
However, Trunk lines T1 and T2 of PBX-A are connected directly to the PSTN.
In such application scenarios, implementing toll control restrictions for the trunks is a difficult task for ETERNITY.
For example: Extension number 21 of ETERNITY in the above illustration is not allowed the facility of long distance
dialing. It has access to all the TWT trunks.
When the user of Extension 21 wants to access T1, T2 or T3 (which are direct trunks from the PSTN to ETERNITY)
the user dials '0' (Trunk Access Code programmed), gets PSTN dial tone. When the user dials the number,
ETERNITY applies Toll Control.
When the user of Extension 21 tries to grab a trunk T5, T6 or T7 (which are connected to stations of PBX-A) by
dialing Trunk Access Code, for example, '0', the user gets the dial tone of PBX A. This means, the user of
Extension 21 must dial '0' again to grab PSTN dial tone of the T1/T2 connected to PBX-A.
But when the user dials '0' again, ETERNITY plays an Error Tone, because ETERNITY has applied Toll Control
and since Extension 21 is not allowed long distance dialing, ETERNITY rejects dialing on trunk and plays error
tone.
The Pre-PSTN Digit Count defines the number of digits to be dialed to reach the PSTN. The system will apply Toll
Control check for the extension only after the programmed PPDC.
PPDC is to be programmed only for trunks that are connected to another PBX, and not for Trunks connected
directly to the PSTN. To take the above illustration further, PPDC must be programmed only for T5, T6, and T7.
PPDC count is to be programmed should have the same number of digits as the Trunk Access Codes programmed
for PBX-A. For example, if the Trunk Access Code is a single digit number, such as '0', the PPDC will be '1'. If Trunk
Access Code is a two-digit number, such as 61, the PPDC will be '2'.
Since PPDC is not applicable on trunks directly connected to the PSTN, it must be programmed as '0' for T1, T2,
T3, T4 of ETERNITY.
How to configure
The 'Pre-PSTN Digit Count' (PPDC) is to be programmed in the “TWT Hardware Template” applied to the TWT
trunks of the PBX that are connected to station ports of the other PBX as well as to TWT trunks that are directly
connected to the PSTN.
• For TWT Trunks that are directly connected to the PSTN, PPDC must be programmed as '0'.
• For TWT Trunks that are connected to the stations of another PBX, PPDC must be programmed as per the
number of digits in the Trunk Access Codes defined for the second PBX.
The PPDC should be programmed only for 'Behind the PBX Applications'. For all normal applications, this
count must be set to '0' for all the trunks. Otherwise, external number dialing may be hampered. Features
like Least Cost Routing and Station Message Detail Recording will also be affected.
• Click 'TWT Hardware Template' under TWT Configuration to open the page.
• By default TWT Hardware Template Number 01 is assigned to all trunks. The default 'PPDC' in this
template is '0'.
• For all trunks that are to be assigned PPDC '0' (i.e. trunks connected directly to the PSTN), you may retain
this template.
• For trunks that are to be assigned a PPDC count from 1 to 6 (i.e. trunks connected to the stations of
another PBX), prepare another TWT Hardware Template by selecting another template number, for
instance Template 02.
• From the drop down list, select the appropriate value. This would depend on the number digits in the Trunk
Access Code defined for the trunks in the other PBX. If the TAC is single digit, select '1'. If TAC is double
or triple digit, select '2' or '3' as applicable as the PPDC.
• Enter the number of the template you prepared (Template 02) in the field "TWT Hardware Template"
for each port you want to assign this template.
• For Trunks to be assigned PPDC Count '0', retain TWT Hardware Template Number 01.
To assign the TWT Hardware Template now programmed with the PPDC to trunks, dial:
• 5903-1-TWT-Template Number to apply the template on a single TWT trunk port.
• 5903-2-TWT-TWT-Template Number to apply the same template on a range of TWT trunk ports.
• 5903-*-Template Number to apply the same template on all TWT trunk ports.
Where,
TWT is the number of the Software port of TWT Trunks, from 001 to 128.
Template Number is TWT Hardware Template number programmed with the PPDC, from 01 to 50.
For instance: To apply Template Number 02 to TWT 003 to 005, dial 5903-2-003-005-02.
• Exit SE Mode.
What's this?
BITE (Built-In Test Equipment) is an automatic testing facility offered by the ETERNITY to test the functioning of
trunk and station ports of the ETERNITY.
BITE is supported on the “The SLT+MAG+TWT+LD+ENM Card”. To be able to use BITE, you must have this card
installed in the system. You can test the functioning of SLT, Magneto, TWT, Loop Dial and E&M ports using BITE.
How it works
• The functioning of a port is tested by looping back the port with the opposite gender port, i.e. the BITE Port.
• The backbone of the system has a dedicated BITE bus. The port being tested is connected with the
opposite gender BITE Port via the BITE bus at the backbone.
• In the 'Normal' mode, the Port is connected with the RJ45 MDF connector of the card.
• When in 'Test' mode (when you conduct a BITE test), the Port is removed from the RJ45 MDF connector
and gets connected to with the BITE Bus at the backbone of the system.
• When the SA command to start BITE is dialed, the system locates opposite gender port BITE port for the
port being tested and loops it back with the BITE port.
• The following tests are performed for the respective trunk/station port being tested when BITE is activated:
Release (On-Hook)
Speech
Release (On-Hook)
Speech
Ring Detection
Outgoing Ring
Speech
Release
Speech
• The port will be tested according to the current configuration of port. For example, if a TWT port being
tested has Pulse as Dial type in its configuration, BITE will check only for Pulse.
• The port under test and the BITE port will remain "Busy" until the test is completed. It will not be possible to
make or receive the calls on port under test as well as BITE port at this time.
• After the BITE test is completed, the tested port and the BITE port will switch back to "Normal" mode. Both
will become 'free' and calls can be made or received.
• A BITE Test will be terminated and considered as failed in the following conditions:
• If there is an ongoing call on the port being tested or the BITE port at the time of activating BITE.
• When any of the tests performed in the BITE test gets failed.
• When all opposite gender BITE ports are in use in BITE card.
• When the port under test does not support BITE functionality.
• When the card present in the BITE Card Slot does not support BITE functionality.
• When either BITE card or card under test gets restarted while BITE test is under process.
• When any BITE test is currently in process and the System Administrator tries to run BITE test for
another port. Here the latest BITE test and not the currently running BITE test will be considered as
failed.
• The results of the BITE test are recorded in the System Activity Log. The result of the BITE test performed
for a port is appears on the log with the Slot Number and Port Offset of the port. A successful test is
indicated as 'Pass' and an unsuccessful test is indicated as 'Fail' in the log.
How to configure
To be able to use BITE, you must define the BITE Card Slot, i.e. the slot in which the “The
SLT+MAG+TWT+LD+ENM Card” is installed. This can be done using Jeeves or a telephone.
• Go to 'BITE Card Slot Number' and select the Slot Number, in which the BITE card is installed. By default,
no slot is selected (00).
• Exit SE mode.
How to use
BITE is activated from SA mode. You can activate BITE from Jeeves as well as using a Telephone.
• Select the Port you want to test. You must select the Slot Number (of the card on which the desired port is
located) and the Port Offset (the number of the port on that card).
• The system will inform you about the time that will be taken to complete the BITE test with the message
"Refresh the page after xx seconds to check the result", below 'Submit".
• To see the results, click 'Refresh' after the number of seconds have elapsed.
• Exit SA mode.
You may view the BITE results on the “System Activity Log”.
What's this?
ETERNITY offers the Building Intercom application as the telecom and security solution for commercial and
residential buildings, such as malls, shopping complexes, residential apartment blocks and gated-communities.
Presently, the Building Intercom application is supported in the ETERNITY ME and GE models and their variants.
The Intercom Line Card is available in the following configurations for the variants of ETERNITY ME and GE.
For installation instructions, under Installing ETERNITY ME, see “The Intercom Line Card” and under Installing
ETERNITY GE, see “The Intercom Line Card”.
Caution: When installing ETERNITY ME/GE for Building Intercom, you can install only the Intercom Line
Cards in the system. When an Intercom Line Card is present in the system, other Trunk cards or Extension
or Combination cards present in the system will not work.
What's this?
The feature Call Back on Trunk Ports is used to respond to missed calls from particular numbers on the different
trunk ports of ETERNITY: Two-wire Trunks, Mobile trunks, BRI trunks, T1E1PRI trunks, and SIP Trunks.
When Call Back feature is enabled on a trunk port, and there is a missed call on that trunk port, the ETERNITY
determines if the calling number is eligible for a call back or not. It calls back the same number or an alternative
number programmed for that number, either from the port on which it was received or from a different port,
depending on the programming. ETERNITY can be programmed to choose the most cost effective line to call back
the missed call numbers.
Employees at remote locations can use this feature to have the ETERNITY installed in their office call them back,
thereby saving on charges (for example, roaming charges on mobile calls), where applicable.
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic “License Management” to know more.
How it works
For this feature to work:
• The CLI of those callers whom the system should call back must be programmed in the ‘Call Back
Incoming Number List’.
• The ‘Call Back Timer’ may be programmed. When the caller disconnects within the Call Back Timer, the
Call Back will be applied for that number.
• You must define ‘Call Back on’, i.e. you must select whether the number which must be called back should
be the same CLI number which the call was received or an alternative number.
• The number on which call back is to be made must be programmed in the ‘Call Back Outgoing Number
List’, if it is not the same CLI number or if it is an alternative number.
• You must select whether the call back should be made using the same trunk port on which the call was
received or an Outgoing Trunk Bundle Group (OGTBG). If you select OGTBG, you must also program the
OGTBG.
• You may enable Least Cost Routing (LCR) on the OGTB if you want the system to select the least cost
trunk for calling back the missed call number. Program LCR accordingly.
• Select a ‘Call Back Mode’, i.e. how the call should be routed when the call back is answered by the remote
party; whether it should be routed as DID, DISA or Operator.
• The system checks if the Call Back flag is enabled on mobile port 01.
• The system matches the CLI of A with the Call Back Incoming Number List assigned to mobile port 01 to
determine if the calling number is eligible for a call back.
• The system waits for the period of the Call Back Timer (programmable, default: 10 seconds).
• A must disconnect before the expiry of the Call Back Timer so that the system can treat it as a Missed Call.
• If A disconnects within the Call Back Timer, the system applies Call Back for A’s number.
• The system checks the ‘Call Back on’ parameter, whether it has to call back the same number or an
alternative number.
• If an alternative number is programmed as ‘Call Back on’, the system checks the Outgoing Call Back
Number List for the alternative number. As the CLI of A matches with the number on Index 15 of the Call
Back Incoming Number List, the system checks Index 15 of the Call Back Outgoing Number List for the
corresponding alternative number to this number.
• The system checks if the number is to be called from the same port or an OGTBG.
• If the same port is programmed, the system will make a call to the number using mobile port 01.
• If OGTBG is programmed, the system will check if Least Cost Routing is enabled in the OGTBG and make
the call back accordingly.
• The system checks the type of Call Back Mode enabled on mobile port 01 (the port on which the call back
request was made).
1. “Direct Inward Dialing (DID)” is enabled as Call Back Mode on mobile port 01.
2. 'Pin Authentication - Multiple Calls' or 'CLI Authentication - Multiple Calls' is enabled as Call Back Mode
on mobile port 01.
128. If the system does not find a match for the CLI of the caller in the Call Back Incoming Number List, the 'Call
Back' feature will not be applicable and the call will be processed according to the normal incoming call logic.
• A can now reach any station or trunk of ETERNITY from DISA Mode.
3. 'CLI Authentication - Single Call Answer Signaling' is enabled as Call Back Mode on mobile port 01.
• The system lands the call on the Operator station assigned to mobile port 01.
Read the topics “Direct Inward Dialing (DID)”, “Direct Inward System Access (DISA)” and “Configuring
'Operator'” to know more about the call respective call logic.
• Since this feature is essentially for callers, they must be aware of its functioning to be able to use it, i.e.
disconnect the call within the Call Back Timer. If the caller does not disconnect within the Call Back
Timer, the call will be processed according to the normal incoming call logic.
• ETERNITY supports only one call back request at a time, for one trunk port. The second incoming call
on that trunk port will be processed by the system as per normal incoming call routing.
• For call back requests made from an OGTBG, if any of its trunks is busy, ETERNITY will support only
the last call back request in the OGTBG. Previous requests will be processed as per the normal
incoming call management logic.
How to configure
For this feature to function, you must program the following parameters on each Trunk port type (TWT, BRI, T1, E1,
Mobile, SIP) on which you want to use this feature:
• Enable Call Back: This flag must be enabled on the desired trunk port on which you want to activate the
Call Back on Trunk Port feature. By default, this flag is disabled on all trunk port types.
• Call Back Timer: This is the duration for which the system waits for the caller to disconnect the call after
the system has found a matching number for the caller’s CLI in the Call Back Incoming Number List.
When the caller disconnects within Call Back Timer, the system applies Call Back on the port. If the caller
does not disconnect within the Call Back Timer, the incoming call management logic is applied for the call
on the trunk port.
• Call Back Incoming Number List: This is the list of numbers that are eligible for Call Back. The system
checks the CLI of the caller with this list to determine if the caller is eligible for a call back.
The system compares the number string programmed in the Call Back Incoming List with the number
string received as CLI.
The number string programmed in the Call Back Incoming Number List may be shorter than the number
string received as CLI, but only if the programmed number string completely matches with the received
CLI from the right towards left, the system will consider it as a complete match.
For example, if the programmed string is 263055 and the number string received in the CLI is
2652630555, the system will consider it a complete match. If the received CLI 912652630555, the system
will consider this caller too as eligible for a call back. Thus any CLI received with 263055 as the last 7 digits
will be considered as match found.
By default, ‘Number List’ 15 is assigned to all trunk port types as Call Back Incoming Number List. You
may program this list for all port types, or you may program another Number List and assign it to the
particular trunk port type.
Refer the topic “Number Lists” for instructions on how to configure the Number List.
• Call Back on: For each Trunk port type you have set the Call Back feature, you must define ‘Call Back on’,
i.e. you must select whether the number which must be called back should be the same number from
which the call was received or a different number.
When missed call is eligible for call back (matches with Incoming Number list), the 'Call Back on'
parameter determines the number on which the call back is to be made, i.e. whether on the same number
from which the missed call is received or on a different number.
In countries where CLI received on trunks can be dialed out without any modification, you may select ‘CLI
Number’ as ‘Call Back on’ option.
In countries where CLI received on trunks can be dialed only after appropriate modification, you may
select “Alternate Number’ as the ‘Call Back on’ option. You may also select ‘Alternate Number’ as Call
Back on when you want the call back to be made to a different number.
• Call Back Outgoing Number List: When the system finds a missed call eligible for a call back, it will
make the call back on the basis of the Call Back on option you selected and the Outgoing Number List you
programmed.
If you selected ‘CLI Number’ as “Call Back on’ option, you do not need to program the corresponding
outgoing number for the CLI received.
However, if the CLI received needs to be modified before being dialed out, then program the modified CLI
in the Outgoing List as the corresponding outgoing number for the CLI received.
The modified CLI or the Alternate number should be programmed at the same index number as the index
number at with the received CLI is programmed in the Call Back Incoming Number List. For example, for
the received CLI number string programmed at Index 15 in the Call Back Incoming Number List, the
corresponding modified CLI/Alternate number string should be programmed at the same Index, 15, in the
Call Back Outgoing Number List.
When the CLI received matches with the number string programmed at Index 15 of the 'Call Back
Incoming Number List', the call back will be made using the (modified/Alternate) number programmed at
Index 15 of the 'Call Back Outgoing Number List'.
Refer the topic “Number Lists” for instructions on how to configure the Number List.
If you have selected ‘Alternate Number’ as ‘Call Back on’ option, but do not want to provide alternative
numbers to call back particular callers (i.e. CLI received), in such a case, program the CLI of these callers in
the Incoming Number List but keep the corresponding index numbers in the Outgoing Number Lists blank.
• Call Back from: This parameter determines the trunk port to be used to make call back.The call back can
be made using the same port or an Outgoing Trunk Bundle Group (OTGTBG). Select ‘Same port’ if you
want the call back to be made using the same port on which the missed call was received. If you select
OGTBG, the call back will be made using the OGTBG, which you have defined.
• OGTBG for Call Back: If you selected OGTBG for making the call back in the previous parameter, you
must assign the OGTBG that must be used in this parameter.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost Routing
on the OGTBG that you define here for Call Back.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote party
should be routed:
• DID: The system will process the call as per the DID call logic - give a dial tone to the remote party,
who can now call any extension. Refer the feature description for “Direct Inward Dialing (DID)”.
• PIN Authentication-Multiple Calls: The system will process the call as per DISA call logic - allow
remote party to enter DISA mode with PIN-Authentication. On successful authentication (DISA
Login) the user is allowed to make calls or use features as allowed to him/her.
• CLI Authentication-Multiple Calls: The system will process the call as per DISA call logic,
allowing the remote party to enter DISA mode with CLI Authentication-Multiple calls as
authentication method and level of access.
• CLI Authentication-Single Call-Answer Signaling: The system will process the call as per DISA
call logic, allowing the remote party to enter DISA mode with CLI Authentication-Single call as
authentication method and level of access. Refer the feature description for “Direct Inward System
Access (DISA)”.
• Operator: When the remote party answers the Call Back call, the system will route the call to the
Operator129.
All these parameters may be programmed using Jeeves or by dialing SE commands from a telephone.
To program Call Back on different port types, refer the relevant topics mentioned below:
129. 'Operator' is the station which is assigned to the Mobile port in the Trunk Feature Template. Refer Trunk Feature Template to know
more.
• For Call Back on Mobile Ports, refer the topic “Configuring Mobile Trunks”.
• For Call Back on BRI Ports, refer the topic ‘BRI Parameters’ under “ISDN-BRI”.
• For Call Back on T1E1 Ports, refer the topic ‘Call Back on T1E1 Trunk Ports’, under “T1E1 Trunks”.
• For Call Back on SIP Trunks, refer the topic “Configuring SIP Trunks”.
What's this?
Call Budget is a cost control feature that allows you to keep a tab on the total cost of phone call made by extension
users.
With this feature, each extension can be allotted a 'budget' limit for outgoing calls, which is automatically reloaded
at the start of every month.
Long distance calls form a major part of the increased cost of telephone calls. Though excessive use or misuse of
long distance dialing can be restricted using Toll Control, there may be extension users whose nature of work
requires them to make long distance calls. Instead of denying them the facility, their telephone bill can be limited to
a certain amount using Call Budget.
With a Call Budget allotted to the extension, the user is free to make calls as long as s/he does not cross the budget
limit. Once the user exceeds the budget limit, the extension can be denied access to long distance dialing.
The extension user can be assigned a fresh budget, after which s/he can resume making long distance calls.
Call Budget can be enabled on all the extensions as well as on selected extensions. Each extension can be
assigned a different amount depending on user requirement.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How it works
When an extension allotted Call Budget makes a call,
• The system checks the current call budget amount of the extension.
• If the consumed amount is within the budget limit allotted to the extension,
• The system allows the extension to make the call as per the “Toll Control Levels” assigned to it.
• After the call ends, the system calculates and adds the call amount to the extension's account. Thus it
calculates and updates the total cost of calls made from the phone.
• If the consumed amount exceeds the budget limit allotted to the extension,
• The system allows the extension to make the call as per the Toll Control-Call Budget Consumed
assigned to the extension.
• After the call ends, the system calculates and adds the call amount to the extension's account.
• Until a new Call Budget is allocated to the extension user, the extension user can make calls only as per
Toll Control assigned for the Call Budget Consumed state.
• If the budget exceeds anytime during the month, and if no fresh budget amount is allotted, the system
allows calls to be made as per the Allowed and Denied List of Toll Control-Call Budget Consumed till the
end of the month. From the 1st day of the following month, the system automatically reloads the budget
amount. The extension can now make calls.
• The Call Budget allotted to extension is valid for one month. The system automatically reloads the budget
at the start of every month.
• The budget amount can be changed or allotted afresh to extensions from the System Administrator (SA)
mode, at any time. The Call Budget allotted by the SA will be reloaded in the following month.
• Call Budget is not based on real time (online) call cost calculation. The ETERNITY calculates the call
cost only after the call has ended.
• So, if the Call Budget allotted to an extension user gets exhausted in the middle of a call, the call will
not disconnected, though the budget is exceeded. To prevent this from occurring, the System Engineer
may program the “Call Duration Control (CDC)” feature.
• Call Budget is dependent on precise Call Cost Calculation. So, SMDR parameters and long distance
codes must be programmed properly to prevent errors in calculation.
• This feature works independent of any Call Accounting Software (CAS) installed with the ETERNITY.
• The ETERNITY will calculate cost of phone calls made by extension phones even when no call budget
is allocated130.
How to configure
The working of this feature is controlled by three parameters: 'Call Budget' flag, 'Toll Control-Call Budget
Consumed' and 'Preset Call Budget Amount'.
In the default Station Basic Feature Template 01 assigned to all stations of the ETERNITY, the Call Budget flag is
disabled and the Toll Control-Call Budget Consumed is set to 'No Calls'.
If Call Budget is to be allowed to all stations, simply enable the flag in the default Station Basic Feature Template
01 and select the Toll Control for Call Budget Consumed state.
However, if Call Budget is to be allowed to selected extensions, then prepare a separate Station Basic Feature
Template with the Call Budget flag enabled and the Toll Control-Call Budget Consumed set. Now, apply this
template on stations that are to be allowed this feature.
The new Call Budget set by the System Engineer will be considered as the Preset Call Budget amount. This
amount will be allocated at the start of every month to all extensions having Call Budget feature in their Station
Basic Feature Template.
Further, the System Administrator (SA) can override the Preset Call Budget amount set by the System Engineer,
and allot call budgets on an extension-by-extension basis. For example: allotting higher amount to extensions of
senior managers, Marketing, Sales, Exports departments, and lower amount to extensions that are less likely to
make long distance calls frequently.
The amount may be greater or lesser than the default amount set by the System Engineer. The Call Budget amount
allotted by the SA will be reloaded at the start of every month on the extension. For instructions refer 'How to Use'
later in this section.
• The amount programmed as Preset Call Budget is to be considered as the local currency.
• At the time of installation, when the SE selects the Region Code (country code) and defaults the
system, the related Currency Code is applied.
• The currency symbol will not be displayed on the Operator's phone, on account of the limited number of
characters that can be displayed.
How to use
Call Budget amount can be allotted to extensions from SA mode only.
OR
OR
What's this?
Call Budget on Trunks is an expense control feature of ETERNITY that allows you to keep track and of the cost of
phone calls made from the different Trunk ports of ETERNITY.
With this feature, each trunk can be allotted a 'budget' limit for outgoing calls. This budget limit can be programmed
to be reloaded manually each time it is exceeded or at a scheduled date, either daily or at a particular date of the
month.
There are three types of Call Budget limit that can be set on the trunks:
• Amount: In this type of Call Budget, a fixed amount is assigned to the trunk. By default the amount of
999999 (to be considered in the local currency) is set as Call Budget Amount on trunks. With Amount-
based Call Budget you can control the actual expense incurred on making calls from a trunk.
• Minutes: In this type of Call Budget, a fixed number of Minutes are assigned to the trunk. By default,
999999 minutes are assigned as Call Budget Minutes on trunks. This type of Call Budget is useful when
the Service Provider offers 'Free' minutes. For example, the Service Provider allows the customer to make
calls for the first 1000 minutes every month. This offer can be availed of by programming Minutes-based
Call Budget on the trunk port.
• Number of Calls: In this type of Call Budget, you can define the maximum number of calls that can be
made from a trunk. By default, the maximum number of Call Budget - Calls is set to 9999 calls on the
trunks. This type of Call Budget is useful when the Service Provider offers a certain number of free calls or
a certain number of free calls for a fixed period. For instance, the Service Provider offers 150 free calls per
month.
With a Call Budget allotted to a trunk, the users can make calls from the trunk as long as the budget limit set for the
trunk (i.e. the Amount or Minutes or the maximum number of Calls) is not crossed. Once the budget limit is
exceeded, the trunk gets disabled automatically and no outgoing calls are allowed to be made from the trunk.
The consumed Budget can be reset, after which it becomes functional again and allows outgoing calls to be made.
The consumed Call Budget can be reset manually, i.e. anytime, as required/desired, or on a scheduled date either
daily or on a particular date of the month.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How it works
Call Budget can be enabled on trunk port types - TWT, Mobile, SIP, BRI, T1E1PRI- all at once or on selected trunk
port types from among them. Each trunk can be assigned a different Call Budget, depending on the requirement of
the users.
When Call Budget is enabled on a trunk port, for each outgoing call,
• The system checks the type of Call Budget set on the trunk - Amount, Minutes or number of Calls.
• At the end of each outgoing call made from the trunk, the system will calculate the cost of the call on the
basis of the Pulse Rate Type programmed. The system will thus calculate the total amount consumed after
the end of each call. Refer the topic "Call Cost Calculation" to know more.
• With the number of Minutes defined, at the end of each call, the system will calculate the duration of the
call on the basis of the units programmed in the Pulse Rate. The system will calculate the consumed
minute on the basis of the duration of the call. Refer the topic "Call Cost Calculation" to know more.
• With the number of calls programmed, the system will maintain a count for the number of matured
outgoing calls made from that trunk port.
• Thus for each matured call, the Number of Calls-Count is incremented, irrespective of the actual duration
of the matured call.
• When the assigned 'cost' or 'minutes' or 'number of calls' assigned to trunk is exhausted, ETERNITY will:
• The consumed Call Budget Amount/Minutes/Calls can be reset manually at any time from the System
Administrator mode or the System Engineer mode or can be programmed to be automatically reset either
daily or on a particular date of the month.
• The current Call Budget Amount/Minutes/Calls limit can be changed from the System Administrator (SA)
mode, at any time. If scheduled reset of consumed Call Budget is programmed, then the Call Budget
allotted by the SA will be reloaded on the scheduled date.
• Once a new Call Budget is allocated to the trunk, outgoing call facility is resumed on the trunk.
• Call Budget on Trunks is not based on real time (online) call cost calculation. The ETERNITY
calculates the call cost only after the call has ended.
• If the Call Budget allotted to a Trunk Port gets exhausted in the middle of a call, the call will not
disconnected, though the budget is exceeded.
• This feature works independent of any Call Accounting Software (CAS) installed with the ETERNITY.
• The ETERNITY will calculate cost of phone calls made by the trunks even when no call budget is
allocated131.
How to configure
Call Budget on Trunks is to be programmed in the Trunk Port Parameters of the trunk type on which you want to
enable this feature. This can be done using Jeeves as well as a telephone
• Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this TWT trunk port:
• Type: Select the type of Call Budget on Trunk, i.e. Amount or Minutes or Calls to be applied on this
TWT trunk port. By default, no Call Budget type is selected.
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
• Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
• Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.
• You may program the same Call Budget parameters as listed above for other trunk types:
• Click 'Mobile Port Parameters' under 'Mobile Configuration' to program Call Budget on Mobile Ports.
Click the 'Advance' button on this page to reach Call Budget parameters.
• The consumed Call Budget on trunk can be reset from the System Engineer mode as well as the
System Administrator mode manually at any time, referred to as Manual Reset.
• Manual Reset of Call Budget on Trunks by the System Engineer can be done either from Jeeves or
using a Telephone.
• Open the 'Status' page under 'TWT Configuration'. Select the 'Reset Consumed Amount/Minutes/Calls'
check box of the TWT port for which you want to reset the consumed Call Budget.
• Similarly, open 'Status' page under 'Mobile Configuration' to enable the same parameter 'Reset
Consumed Amount/Minutes/Calls' of the Mobile port for which you want to reset the consumed Call
Budget.
• To manually reset the consumed Call Budget on T1E1PRI trunks, open 'Status' page under 'T1E1
Configuration'. Enable the parameter 'Reset Consumed Amount/Minutes/Calls' for the T1E1 port for
which you want to reset the consumed Call Budget.
• To manually reset the consumed Call Budget on BRI trunks, open 'Status' page under 'BRI
Configuration', and enable the parameter 'Reset Consumed Amount/Minutes/Calls' for the BRI port for
which you want to reset the consumed Call Budget.
• Open 'Status' page under 'VoIP Configuration' and enable the parameter 'Reset Consumed Amount/
Minutes/Calls' to manually reset consumed Call Budget on the desired SIP trunks.
To program Call Budget Reset Mode for Mobile trunk port, dial:
• 8022-1-Mobile-Call Budget Reset Mode to program reset mode for a single trunk port.
• 8022-2-Mobile-Mobile-Call Budget Reset Mode to program the same reset mode for a range of trunk
ports.
• 8022-*-Call Budget Reset Mode to program the same reset mode for all trunk ports.
Where,
Mobile is the number of the Mobile software port from 01 to 64.
1 for Scheduled reset
2 for Manual reset
By default, Call Budget Reset Mode is Scheduled.
To program the Date for Scheduled Reset mode for Mobile trunk port, dial:
• 8023-1-Mobile-Date to program reset date for a single trunk port.
• 8023-2-Mobile-Mobile-Date to program the same reset date for a range of trunk ports.
• 8023-*-Date to program the same reset date for all trunk ports.
Where,
Mobile is the number of the Mobile software port from 01 to 64.
Date is
01 to 31 for Scheduled date to reset every month.
00 for Scheduled reset Daily.
By default, Reset date is 1st. of every month.
To program the Date for Scheduled Reset mode for BRI, dial:
• 6218-1-BRI-Date to program date for a single trunk.
• 6218-2-BRI-BRI-Date to program the same date for a range or trunks.
• 6218-*-Date to program the same date for all trunks.
Where,
BRI is the number of the BRI software port from 01 to 32.
Date is
01 to 31 for Scheduled date to reset every month.
00 for Scheduled reset Daily.
By default, Reset date is 1st. of every month.
• Exit SE mode.
• Exit SE mode.
What's this?
Call Chaining is when an external/internal call transferred by the Operator to another extension or external number
is made to return to the Operator's extension after the conversation between the caller and the extension/external
number to which it is transferred has ended.
Call Chaining is useful situations where the Operator intervention is required after the transferred call has ended.
For instance:
• The caller needs to take an appointment or requires some information from the Operator after talking to the
desired extension.
• A marketing executive who calls his supervisor to consult on a technical problem needs to be informed
about his travel itinerary and ticket booking by the Operator. The Operator can transfer the call to the
supervisor, and use Call Chaining to retrieve the call once the conversation has ended to give the
information to the executive.
How it works
• A is an External Caller
• B is an extension.
• If A disconnects the call with C, the call will be released. It will not return to the Operator.
• If the Operator is busy, A will be played music on hold for the duration of the Call Park Release Timer.
• If the Operator is busy and the Timer elapses, the call will be released.
Call Chaining can be performed when call is transferred from a DKP to another DKP, SLT or Trunk.
The process of Call Chaining would be the same if A were an internal caller and B were an external number,
or if both were external numbers or both were internal numbers.
Refer the topic “DSS Keys Programming” and “Call Park” for instructions.
How to use
OR
• Press Hold.
• Dial 1050.
• Caller gets on-hold music.
• You get confirmation tone and the message 'Called Party in Chaining' on your phone's display. If DSS Key
is used, the LED of the key will glow.
• Dial the requested extension/external number.
• You get Ring Back Tone.
• The called party answers.
• Perform Call Transfer132 to the requested extension/trunk.
• When extension/trunk disconnects, your extension rings.
• Go OFF Hook. You get connected with the caller.
• Go Idle after the conversation ends, or you get dial tone after 3 seconds.
132. Refer the EON48 and EON42 User Cards for instructions on various options for Call Transfer of internal and external calls.
What’s this?
• The ETERNITY can calculate the cost in amount for the calls made by the station. The cost calculation is
done at the time of printing of reports.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How it works
Few parameters are programmed to calculate the cost of the call. The cost of a call depends on:
• Time and day when the call was made i.e. daytime, nighttime, holiday, etc.
The ETERNITY will calculate the 'call cost' by using following steps:
• When the call is made, the 'Call Cost Calculation Pulse Rate Option' is checked for the specific trunk,
based on the 'Call Cost Calculation Time Schedule' for the OG call. The pulse rate option (from 1 to 4), is
as assigned in Trunk Feature Template. Refer chapter “Trunk Feature Template”.
• The 'Number' dialed will be matched with 'Area Code Table' programmed in ETERNITY. For the matched
area code and, the 'Pulse rate type' programmed for the 'Pulse rate option' is obtained.
• This 'Pulse Rate type' obtained from 'Area Code Table' will be checked in 'Pulse Rate table' (The Table
may be for 'Normal' or 'Holiday', depending on the day of call) to obtain corresponding 'duration' and 'cost'
to be applied for the call duration.
• The 'pulse rate type' applied (duration and cost) is divided into two parts for each time zone:
• First unit.
• Additional units.
The duration of the call is interpreted in terms of number of units and the number of units depends on the pulse
rate.
ETERNITY will use the 'cost of the call' for SMDR and also to deduct it from the call budget, if allotted to trunk.
Example:
• Suppose, OG call is made on trunk, TWT-001 which is assigned Trunk Feature Template number '01'.
001 26 Local 03 06 09 10
002 09 Mobile 05 03 07 08
Duration (sec) 300 300 300 300 300 300 300 300
02
Cost 01.00 01.00 01.00 01.00 01.00 01.00 01.00 01.00
Duration (sec) 30 30 30 30 30 30 30 30
03
Cost 01.00 01.00 01.00 01.00 01.00 01.00 01.00 01.00
Duration (sec) 45 45 45 45 45 45 45 45
04
Cost 01.00 01.00 01.00 01.00 01.00 01.00 01.00 01.00
Duration (sec) 180 180 180 180 180 180 180 180
05
Cost 03.00 03.00 03.00 03.00 03.00 03.00 03.00 03.00
Duration (sec) : : : : : : : :
:
Cost : : : : : : : :
Duration (sec)
32
Cost
• Suppose OG call is made by the station user, to number "2630555" through the trunk, TWT-001 at 20:10
hours. The ETERNITY will check trunk feature template assigned to TWT-001 and determine Time Zone
as per time of the call. The corresponding Pulse Rate Option configured in Trunk feature template will be
checked.
In this example, Time Zone for TWT-001 at 20:10 Hours would be T1.
Pulse Rate Option is = 1.
• The ETERNITY will match the dialed number "2630555" in Area Code table, in which, the entry
programmed at index 001 is found as best match.
Hence, as per Area Code Table, pulse rate type = 03, programmed in "Pulse Rate Option -1" for matching
entry (Index 001).
(However, if TWT would have been assigned Pulse Rate Option=2, in trunk feature template, the pulse
Rate type 06 would have been selected as shown in Area Code Table)
• Finally, for Pulse Rate Type = 03, ETERNITY will consider Cost for First Unit as 1.00 (Rs. or $ as per
applicable currency) for duration of 30 seconds and for additional unit also cost will be considered as 1.00
for duration of 30 seconds. (As shown in Pulse Rate Table). This data will be used for calculating total cost
of call based on the total duration.
How to configure
Step 1
Unit Charge
When call cost is to be calculated on the basis of 16 KHz metering pulses, user can select different unit charge for
first unit and different unit charge for the additional units.
Use following command to program the unit charge for first unit when 16 KHz metering is used:
2600-Unit Charge for First Unit
Where,
Unit charge is the amount in XX.XX format in any currency.
By default, unit charge for first unit is Rs.1.10.
Use following command to program the unit charge for additional units when 12/16 KHz metering is used:
2601-Unit Charge for Additional Unit
Where,
Example1:
Let us program unit change for first unit to Rs. 1.50.
2600-0150
Example2:
Let us program unit charge for additional unit to US$0.75.
2601-0075
Step 2
Service Charge
• Fixed Service Charge: A fixed amount is added as service charge to every call regardless of the cost of
that call. This service charge amount is programmable.
• Unit Wise Service Charge: Service charge is added to each unit of the call. Suppose a call worth 10 units
was made then the service charge will also be charged for 10 units instead of once as it is done in case of
fixed service charge.
• Percentage Wise Service Charge: A percent of the cost of the call is added as a service charge for that
call. This percent is programmable.
Step 3
Use following command to program service charge:
2603-Service Charge
Where,
Service Charge is the amount in XX.XX format in any currency.
By default, Service Charge is Rs.00.00.
Example 1:
Let us program the service charge to Rs.2.00.
2603-0200
Example 2:
Let us program the service charge to US$1.75.
2603-0175
Example:
Let us set the percentage to 10% of the cost of each call.
2604-010
Number of Units
• Number of Units is derived from the pulse rate at the time of the call and duration of the call. System
acquires the pulse rate type and call duration with the help of in-built RTC.
• If the call duration is less than the pulse rate of the first unit then additional unit is zero and total units. Call
units when call answer supervision type is 12/16KHz metering.
Step 5
Assign parameters; 'CCC Pulse Rate Option' and program 'CCC Time Schedule' for the Trunk Feature Template
which is assigned to the specific trunk used for OG calls.
• Program four 'CCC Time Schedule', T1, T2,T3 and T4. Program Start Time and End Time for each.
Step 6
Pulse rate can differ for normal and holidays. Maximum 32 entries can be made in the pulse rate type. Each pulse
rate type can have different rate and different cost for first and additional unit. The table below shows the format of
pulse rate for normal day.
Duration : : : : : : : :
02
Cost : : : : : : : :
: : : : : : : : : :
Use following command to program duration of first unit for a pulse rate type on normal days:
2607-Pulse Rate Type-Time Zone-Duration of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Duration of First Unit is from 000.00 to 999.99.
Step 7
Use following command to load default normal pulse rate type:
2606
Step 8
Use following command to program duration of additional unit for a pulse rate type on normal days:
2608-Pulse Rate Type-Time Zone-Duration of Additional Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Duration of Additional Unit is from 000.00 to 999.99.
Duration : : : : : : : :
:
Cost : : : : : : : :
Step 9
Use following command to program the cost of first unit of a pulse rate type for normal days:
2609-Pulse Rate Type-Time Zone-Cost of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of first Unit is from XX.XX.
Step 10
Use following command to program the cost of additional unit of a pulse rate type for normal days:
2610-Pulse Rate Type-Time Zone-Cost of Additional Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of Additional Unit is from XX.XX.
Separate pulse rate table is used for holidays. This allows total Flexibility of rates for holidays.
Step 11
Use following command to load default holiday pulse rate type:
Duration : : : : : : : :
:
Cost : : : : : : : :
Step 12
Use following command to program duration for a first unit of a pulse rate type on holidays:
2612-Pulse Rate Type-Time Zone-Duration of First Unit
Where,
Step 13
Use following command to program duration for additional unit for a pulse rate type on holidays:
2613-Pulse Rate Type-Time Zone-Duration of Additional Unit
Where,
Pulse rate type is from 01 to 32.
Time Zone from 1 to 4.
Duration of Additional Unit is from 000.00 to 999.99.
Step 14
Use following command to program the cost of first unit of a pulse rate type for holidays:
2614-Pulse Rate Type-Time Zone-Cost of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of First Unit is from XX.XX.
Step 15
Use following command to program the cost of additional unit of a pulse rate type for holidays:
2615-Pulse Rate Type-Time Zone-Cost of Additional Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of Additional Unit is from XX.XX.
• The pulse rate of a call depends on the destination number dialed. Generally, pulse rate varies depending
on the distance. Different destination locations can have different pulse rates depending on the distance
from the caller. Hence pulses rates can vary with different area codes. The Area Code Table in ETERNITY
will be programmed with parameters shown below:
001
002
003
004
999
For example, pulse rate type for different area codes can be programmed as:
Destination Location Area Code Pulse Rate Type for Pulse Rate Option-1
USA 001 15
Japan 0071 14
Singapore 0065 13
Southern Region 04 08
Mumbai 022 06
Rajkot 0281 05
Waghodia 952668 04
Halol 95 03
Surat 95261 04
Ahmedabad 9579 00
The 'pulse rate type' is retained even during power failure conditions. The Area Code table can be printed
or can be downloaded on a computer for reference. Refer “Default Area Code Table for the Region-USA”
at the end of chapter.
• Sort the area codes in increasing order. Write applicable rate codes against each area code.
• Select area codes having common digits (2, 3, 4 or more) with same pulse rate.
• Select this group of codes and find out maximum common digits.
• Use this truncated area code as a common area code for all the area codes in the group.
• Program this common area code with the common pulse rate for the whole group.
• Repeat this procedure to find out other such groups with common pulse rates.
• The codes, which do not belong to any group, should be entered individually as separate entries.
• Usually, area codes of places in the remote area from your city can be compressed in one common code.
01 0111 03
02 0112 03
03 0113 03
04 0114 03
05 0115 03
06 0116 03
07 0117 03
08 0118 03
09 0119 03
10 0110 03
Note that all the above area codes have same pulse rates. Hence it is possible to compress them to one code: 011.
You can program this code with the common pulse rate 03. Thus, one entry in the system area code table will cover
10 entries.
The compressed area code table would look as shown below:
1 0111
Example 2
01 0131 03
02 0132 03
03 0133 03
04 0134 03
05 0135 03
06 0136 03
07 01372 03
08 01374 03
09 01376 03
10 0138 03
11 0139 03
01 013 03
02 0137 03
03 01372 03
Refer “Default Area Code Table for the Region-USA” at the end of chapter.
Step 16
Use following command to program an area code:
2620-Area Code Index-Area Code-#*
Where,
Area Code Index is from 001 to 999.
Area Code is a number string of maximum of 4 digits.
Use the following command to clear the area code for an index:
2620-Area Code Index-#*
Example:
Program area code 022 for Mumbai at area code index 001.
2620-001-022-#*
Step 17
Use following command to program Pulse Rate Type for Pulse Rate Option of area code index:
2621-Area Code Index-Pulse Rate Option-Pulse Rate Type
Where,
Area Code Index is from 001 to 999.
Pulse Rate Option is from 1 to 4.
Pulse Rate Type is from 01 to 32.
Step 18
Use following command to delete the complete Area Code Table:
2622-Reverse SE Password
Example:
To delete the area code table use command:
2622-4321 (The SE password is assumed to be 1234).
There is no command to delete a single entry from the area code table. However, these can be cleared or
overwritten.
Step 19
Use following command to program ignore digit count when SP_SP LCR is to be used:
2623-Area Code Index-Ignore Digit Count
Where,
Area Code Index is from 001 to 999. Refer “Default Area Code Table for the Region-USA”’ at end of the chapter.
Ignore Digit Count is from 0 to 9.
By default, Ignore digit is 0.
Holidays
Step 20
Use following command to program a weekly off:
2630-Day-Code
Where,
Holiday Index 1 2 3 4 5 6 7
Code Meaning
0 Not a Holiday
1 Holiday
If any week day is programmed as holiday then holiday pulse rates will be applicable.
By default, Sunday is programmed as holiday.
Example:
To program Tuesday is a holiday:
2630-3-1
Step 21
Use following command to program a holiday date:
2631-Holiday Date Index-Date-Month
Where,
Holiday Date Index is from 1 to 5 (Five dates can be programmed).
Date is from 01 to 31.
Month is from 01 to 12.
By default, holiday is shown below:
1 26-01
2 15-08
3 02-10
4 Blank
5 Blank
Example:
To program 1st May as a holiday as use following command:
2631-1-01-5
Index Area Code Area Name Pulse Rate Type Ignore Digit Count
1 1201 NJ 2 0
2 1202 DC 2 0
3 1203 CT 2 0
4 1204 Manitoba 2 0
5 1205 AL 2 0
6 1206 WA 2 0
7 1207 ME 2 0
8 1208 ID 2 0
9 1209 CA 2 0
10 1210 TX 2 0
11 1212 NY 2 0
12 1213 CA 2 0
13 1214 TX 2 0
14 1215 PA 2 0
15 1216 OH 2 0
16 1217 IL 2 0
17 1218 MN 2 0
18 1219 IN 2 0
19 1224 IL 2 0
20 1225 LA 2 0
21 1226 Ontario 2 0
22 1228 MS 2 0
23 1229 GA 2 0
24 1231 MI 2 0
25 1234 OH 2 0
26 1239 FL 2 0
27 1240 MD 2 0
28 1242 Bahamas 2 0
29 1246 Barbados 2 0
30 1248 MI 2 0
31 1250 BC 2 0
32 1251 AL 2 0
33 1252 NC 2 0
34 1253 WA 2 0
35 1254 TX 2 0
36 1256 AL 2 0
37 1260 IN 2 0
38 1262 WI 2 0
39 1264 Anguilla 2 0
40 1267 PA 2 0
41 1268 Antigua 2 0
42 1269 MI 2 0
43 1270 KY 2 0
44 1276 VA 2 0
45 1281 TX 2 0
46 1284 BVI 2 0
47 1289 Ontario 2 0
48 1301 MD 2 0
49 1302 DE 2 0
50 1303 CO 2 0
51 1304 WV 2 0
52 1305 FL 2 0
53 1306 Saskatchewan 2 0
54 1307 WY 2 0
55 1308 NE 2 0
56 1309 IL 2 0
57 1310 CA 2 0
58 1312 IL 2 0
59 1313 MI 2 0
60 1314 MO 2 0
61 1315 NY 2 0
62 1316 KS 2 0
63 1317 IN 2 0
64 1318 LA 2 0
65 1319 IA 2 0
66 1320 MN 2 0
67 1321 FL 2 0
68 1323 CA 2 0
69 1325 TX 2 0
70 1330 OH 2 0
71 1331 IL 2 0
72 1334 AL 2 0
73 1336 NC 2 0
74 1337 LA 2 0
75 1339 MA 2 0
76 1340 USVI 2 0
77 1345 Cayman 2 0
78 1347 NY 2 0
79 1351 MA 2 0
80 1352 FL 2 0
81 1360 WA 2 0
82 1361 TX 2 0
83 1386 FL 2 0
84 1401 RI 2 0
85 1402 NE 2 0
86 1403 Alberta 2 0
87 1404 GA 2 0
88 1405 OK 2 0
89 1406 MT 2 0
90 1407 FL 2 0
91 1408 CA 2 0
92 1409 TX 2 0
93 1410 MD 2 0
94 1412 PA 2 0
95 1413 MA 2 0
96 1414 WI 2 0
97 1415 CA 2 0
98 1416 Ontario 2 0
99 1417 MO 2 0
101 1419 OH 2 0
102 1423 TN 2 0
103 1424 CA 2 0
104 1425 WA 2 0
105 1430 TX 2 0
106 1432 TX 2 0
107 1434 VA 2 0
108 1435 UT 2 0
110 1440 OH 2 0
112 1443 MD 2 0
115 1469 TX 2 0
117 1478 GA 2 0
118 1479 AR 2 0
119 1480 AZ 2 0
120 1484 PA 2 0
122 1501 AR 2 0
123 1502 KY 2 0
124 1503 OR 2 0
125 1504 LA 2 0
126 1505 NM 2 0
128 1507 MN 2 0
129 1508 MA 2 0
130 1509 WA 2 0
131 1510 CA 2 0
132 1512 TX 2 0
133 1513 OH 2 0
135 1515 IA 2 0
136 1516 NY 2 0
137 1517 MI 2 0
138 1518 NY 2 0
140 1520 AZ 2 0
141 1530 CA 2 0
142 1540 VA 2 0
143 1541 OR 2 0
144 1551 NJ 2 0
145 1559 CA 2 0
146 1561 FL 2 0
147 1562 CA 2 0
148 1563 IA 2 0
149 1567 OH 2 0
150 1570 PA 2 0
151 1571 VA 2 0
152 1573 MO 2 0
153 1574 IN 2 0
154 1575 NM 2 0
155 1580 OK 2 0
156 1585 NY 2 0
157 1586 MI 2 0
159 1601 MS 2 0
160 1602 AZ 2 0
161 1603 NH 2 0
162 1604 BC 2 0
163 1605 SD 2 0
164 1606 KY 2 0
165 1607 NY 2 0
166 1608 WI 2 0
167 1609 NJ 2 0
168 1610 PA 2 0
169 1612 MN 2 0
171 1614 OH 2 0
172 1615 TN 2 0
173 1616 MI 2 0
174 1617 MA 2 0
175 1618 IL 2 0
176 1619 CA 2 0
177 1620 KS 2 0
178 1623 AZ 2 0
179 1626 CA 2 0
180 1630 IL 2 0
181 1631 NY 2 0
182 1636 MO 2 0
183 1641 IA 2 0
184 1646 NY 2 0
187 1650 CA 2 0
188 1651 MN 2 0
189 1660 MO 2 0
190 1661 CA 2 0
191 1662 MS 2 0
194 1671 GU 2 0
195 1678 GA 2 0
196 1682 TX 2 0
197 1684 AS 2 0
199 1701 ND 2 0
200 1702 NV 2 0
201 1703 VA 2 0
202 1704 NC 2 0
204 1706 GA 2 0
205 1707 CA 2 0
206 1708 IL 2 0
208 1710 US 2 0
209 1712 IA 2 0
210 1713 TX 2 0
211 1714 CA 2 0
212 1715 WI 2 0
213 1716 NY 2 0
214 1717 PA 2 0
215 1718 NY 2 0
216 1719 CO 2 0
217 1720 CO 2 0
218 1724 PA 2 0
219 1727 FL 2 0
220 1731 TN 2 0
221 1732 NJ 2 0
222 1734 MI 2 0
223 1740 OH 2 0
224 1754 FL 2 0
225 1757 VA 2 0
227 1760 CA 2 0
228 1762 GA 2 0
229 1763 MN 2 0
230 1765 IN 2 0
232 1769 MS 2 0
233 1770 GA 2 0
234 1772 FL 2 0
235 1773 IL 2 0
236 1774 MA 2 0
237 1775 NV 2 0
238 1778 BC 2 0
239 1779 IL 2 0
241 1781 MA 2 0
243 1785 KS 2 0
244 1786 FL 2 0
247 1801 UT 2 0
248 1802 VT 2 0
249 1803 SC 2 0
250 1804 VA 2 0
251 1805 CA 2 0
252 1806 TX 2 0
254 1808 HI 2 0
256 1810 MI 2 0
257 1812 IN 2 0
258 1813 FL 2 0
259 1814 PA 2 0
260 1815 IL 2 0
261 1816 MO 2 0
262 1817 TX 2 0
263 1818 CA 2 0
265 1828 NC 2 0
267 1830 TX 2 0
268 1831 CA 2 0
269 1832 TX 2 0
270 1843 SC 2 0
271 1845 NY 2 0
272 1847 IL 2 0
273 1848 NJ 2 0
274 1850 FL 2 0
275 1856 NJ 2 0
276 1857 MA 2 0
277 1858 CA 2 0
278 1859 KY 2 0
279 1860 CT 2 0
280 1862 NJ 2 0
281 1863 FL 2 0
282 1864 SC 2 0
283 1865 TN 2 0
288 1870 AR 2 0
291 1878 PA 2 0
294 1901 TN 2 0
296 1903 TX 2 0
297 1904 FL 2 0
299 1906 MI 2 0
300 1907 AK 2 0
301 1908 NJ 2 0
302 1909 CA 2 0
303 1910 NC 2 0
304 1912 GA 2 0
305 1913 KS 2 0
306 1914 NY 2 0
307 1915 TX 2 0
308 1916 CA 2 0
309 1917 NY 2 0
310 1918 OK 2 0
311 1919 NC 2 0
312 1920 WI 2 0
313 1925 CA 2 0
314 1928 AZ 2 0
315 1931 TN 2 0
316 1936 TX 2 0
317 1937 OH 2 0
319 1940 TX 2 0
320 1941 FL 2 0
321 1947 MI 2 0
322 1949 CA 2 0
323 1951 CA 2 0
324 1952 MN 2 0
325 1954 FL 2 0
326 1956 TX 2 0
327 1970 CO 2 0
328 1971 OR 2 0
329 1972 TX 2 0
330 1973 NJ 2 0
331 1978 MA 2 0
332 1979 TX 2 0
333 1980 NC 2 0
334 1985 LA 2 0
335 1989 MI 2 0
349 01144 UK 2
367 01164 NZ 2
543 2
544 2
545 2
546 2
547 2
: 2
998 2
999 2
Relevant Topics:
1. “Call Cost Display” 917
2. “Call Budget” 870
3. “QSIG” 1402
4. “Configuring LCR” 716
5. “Call Budget on Trunk” 874
6. “Trunk Feature Template” 546
What is this?
With Call Cost Display, you can view the cost of the last 10 external calls made from your extension. These external
calls may have been made from Trunk Ports and on the Tie line network.
The system will display the dialed numbers and the call cost for each number that it has calculated on the LCD of
EON.
How to configure
For this feature to work, it must be enabled on the extension by the System Administrator (SA).
• Enter SA mode.
• Exit SA mode.
How to use
OR
• Dial 1075.
• Scroll with the up/down navigation keys to view the cost of the last 10 calls.
• The display shows the last 10 dialed numbers and their corresponding call cost.
For example: If the call charge is for the dialed number 0014034545247 is $2 and 80 cents' then the
display will show:
What's this?
Call Duration Control (CDC) allows a maximum time limit to be set on internal and external (both incoming and
outgoing) telephone calls. When the maximum call duration is reached, the calls are disconnected, after a warning
tone indicating to the user that the calls in progress will be disconnected.
By limiting the duration of the conversations, CDC helps increase availability of trunks for making outgoing calls
and for receiving incoming calls, which is important in high call traffic situations. Besides increasing trunk
availability, CDC curbs unrelated and unproductive conversations.
How it works
• A is an extension user. B is an external number.
External-Outgoing Calls
• A dials B's number.
• It checks whether the flag, Apply CDC to Outgoing Calls, is enabled. It matches B's number with the
entries on the Apply CDC to Number List and the Do Not Apply CDC to Number list in the CDC table.
Three results are possible:
a. The flag is enabled and a match is found for the number in the Apply CDC to Number List. So, CDC is
applied on the call.
b. The flag is enabled and a match is found in the Do Not Apply CDC to Number List. CDC is not applied
on the call.
c. The flag is enabled and a match is found in both Number Lists, i.e. Apply CDC and Do Not Apply CDC.
The system gives precedence to the Do Not Apply Number List. So, CDC is not applied to the call.
• When CDC is applied to the call (see point a above), the CDC Timer starts as soon as B has answered the
call. This timer is set to 160 seconds as default, but can be programmed to the desired time limit.
• At the end of the default/programmed time limit of the CDC Timer, the CDC Goodbye Timer starts. This
timer provides a grace period of 20 seconds for the user to finish the call. This Timer is non-programmable.
• At the end of the Goodbye Timer, the call is disconnected, if the 'Disconnect CDC after Timer' flag is
enabled.
• If this flag has not been programmed, the call will not be disconnected.
• Instead, the CDC Warn Timer will be loaded again for the default/programmed duration. The user can
know how long s/he has been talking.
External-Incoming Calls
CDC works similarly for incoming calls.
• B calls A.
• The system checks whether the flag, Apply CDC to Incoming Calls, is enabled and matches B's number
with the entries on the Apply CDC to Number List in the table. If the flag is enabled and a match is found
for the number, CDC is applied on the call.
Internal Calls
A and B are extension users.
• A calls B.
• The system checks whether the flag, Apply CDC to Internal Calls, is enabled in the CDC Table.
Warning Beep
Feature Interactions:
• Call Transfer: In case of transferred call, the CDC timer gets reset and starts again afresh on the
transferred extension.
• Emergency Number Dialing: Emergency calls are not affected by this feature, i.e. CDC will not be
applied on the dialing of Emergency Numbers.
• For Inter PINX or Intra PINX calls (QSIG Calls), the CDC will work only if it is enabled on the source port
(calling extension) irrespective of whether CDC is enabled or disabled on the called extension.
• decide the types of calls - Outgoing, Incoming and Internal - on which CDC is to be enabled.
• make a list of numbers on which CDC is to be applied, i.e. the Apply CDC to Numbers List.
• Make a list of numbers on which CDC is not to be applied, the Do Not Apply CDC to Number List.
The Call Duration Control Table can be programmed using Jeeves and a Telephone.
• The CDC Table will open. There are 8 CDC Tables. By default CDC Table No. 1 is assigned to all
extensions of ETERNITY. If the same CDC is to be assigned to all extensions, program this table.
If different CDC is to be applied to different extensions, program separate CDC tables for these
extensions.
• Apply CDC to Internal Calls: This flag is to be enabled if CDC is to be applied on internal calls. By
default the flag is disabled.
• Apply CDC to Incoming Calls: This flag is to be enabled if CDC is to be applied to incoming calls
external calls. By default this flag is disabled.
• Apply CDC to Outgoing Calls: This flag is to be enabled if CDC is to be applied to outgoing external
calls. By default this flag is disabled.
• Do Not Apply CDC to Number List: This is the list of numbers on which CDC is not to be applied. By
default, Number List 08 is assigned to this parameter. You must program this list with numbers which
you want to be exempt from CDC.
To program the list, click 'Do Not Apply CDC to Number List'.
By default, Number List 08 is assigned to this parameter. You can also program any other Number List
you want. Enter the list of numbers on which CDC is not be applied (refer to the list you prepared).
Click 'Submit' at the bottom of the page to save your list.
Return to the "Call Duration Control' page. If you have prepared a Number List other than the default
08, then enter the list of that number in the Do Not Apply CDC to Number List column.
• Apply CDC to Number List: This is the list of numbers on which CDC is to be applied. By default,
Number List 07 is assigned to this parameter. You must program this list with numbers on which you
want CDC to be applied.
To program the list, click 'Apply CDC to Number List'. The 'Number Lists' page opens.
Click '001-250' of Number list 07-08. Follow the same steps as described above for programming the
Do Not Apply CDC Number List.
• CDC Timer: This is the time for which the warning beeps are to be played before the system
disconnects the call. The range of the timer is 0001 to 9999 seconds. By default this Timer is set to 160
seconds. Set the CDC Timer to the desired time limit.
• Disconnect Call after CDC Timer: This flag is to be enabled if you want the call to be automatically
disconnected on the expiry of the CDC Timer. By default the flag is disabled, which means that calls will
not be disconnected on expiry of the CDC Timer. Enable the flag if required.
• To assign the Station Advanced Feature Template with the CDC Table on SLT, DKP and ISDN Terminal
extensions, go to the respective pages 'SLT Parameters' under 'SLT Configuration', 'DKP Parameters'
under 'DKP Configuration' and 'ISDN Terminal Parameters'.
Refer “Station Advanced Feature Template” for instructions on customizing the templates and assigning
them to extensions.
• If selected extensions are to be allowed CDC or if different CDC parameters are to be allowed to selected
extensions (for example, 160 seconds duration timer for a few extensions, 360 duration timer for some
other extensions), then follow these steps:
b. Program the different CDC parameters in this table, as required for the extensions.
d. Apply the new Station Advanced Feature Template now programmed with a different CDC table on the
selected extensions which are to be allowed this feature.
Example: Apply Call Duration Control on SLT 202 (connected at software port number 001) to disconnect all
calls starting with '0' after 240 seconds, except calls starting with '022'.
Solution: Since only one SLT is to be programmed, it is recommended that you use a CDC table other than
the default (CDC Table No. 1). So, select another CDC table to be assigned to SLT extension 202, for
example CDC Table No. 5. Follow these steps:
1. First, program the Apply CDC to Number List and Do Not Apply CDC to Number List. For example, take
Number List 04 as the Apply CDC to Number List and program the number '0' in this list. Take Number List
05 as the Do Not Apply CDC to Number List and program '022' in this list. Refer the topic "Number Lists"
for programming instructions.
2. Enable CDC for Outgoing call in the table. If using SE Commands, dial 4202-1-5-1.
3. Assign Number List 04 as allowed list and Number List 05 as denied list in table 5. If using SE commands
dial 4205-1-5-04 to assign List 04 and dial 4206-1-5-05 to assign List 05.
4. Change the CDC timer to 240 seconds. If using SE Commands, dial 4207-1-5-240.
5. Enable CDC disconnection Flag in CDC Table 5. If using SE Commands, dial 4208-1-5-1.
7. Program a different Station Advanced Feature Template for SLT 202. Ensure that all other features and
parameters in the template are relevant for SLT 202.
9. Assign the Station Advanced Feature Template now programmed with CDC Table No. 5 to SLT 202. Refer
the topic “Station Advanced Feature Template” for programming instructions.
What's this?
By invoking Call Duration Display, extension users can view the duration of the current call instantly.
Needless to say, only EON users can view Call Duration Display. The system displays the duration of the current
call on the LCD display of the phone.
How it works
• The EON user goes OFF-Hook
• The dialed external number with duration (5-digits in the format of MM:SS) is displayed on the LCD of
EON, when the call is answered.
6 1 6 A M I T PAT E L 02:52
F ri 22 JAN 12:19
What's this?
During a typical workday, it is common for people in an organization to move from one place to another. For
instance, a manager might go on the production floor or remain in the conference room for a few hours; a field
engineer may spend half of the day on site. So, they need to be able to attend their calls even when they are not
present at their desks. The 'Call Forward' feature of ETERNITY ensures this.
Using this feature, calls landing on an extension can be forwarded to another extension, an external number, Voice
Mail, or a Department Group. This way, extension users can ensure that callers can reach them and that they do
not miss calls when they are not present at their extension.
The Call Forward feature of ETERNITY offers the following forwarding options:
• Unconditionally - calls are forwarded to the destination phone number automatically without waiting for a
response from the called party's phone.
• If Busy - calls are forwarded to the destination phone number only when the called party's phone is busy.
• If No Reply - calls are forwarded to the destination phone number only when the called party does not
answer the phone. Each extension can set a different time after which the call should be forwarded, in
case of no reply. The default time is 30 seconds for all extensions and can be changed by programming
the Call Forward No-Reply Timer.
• If Busy or No Reply - calls are forwarded to the destination phone number when the called party's phone
is either busy or does not reply.
• Dual Ring134 - when calls are forwarded to another phone number. Both phones, i.e. the source phone
(whose calls are forwarded) as well as the destination phone (on which call is forwarded) will ring and the
user can answer from either extension.
Dual Ring is useful to users who may have to be present frequently at two different places. As it is
cumbersome to forward the calls from one extension to another and cancel it repeatedly, extensions users
can set Dual Ring, so they can attend to their calls at either place they are present.
How it works
A has set Call Forward to extension B unconditionally.
• The system forwards all calls for A to B, without checking for Busy Tone and without waiting for the Call
Forward No-Reply Timer to expire.
• The system waits for the Call Forward No-Reply Timer to expire and forwards all external incoming calls to
the external number.
• The system forwards the call for A to B on detecting Busy signal from A.
B belongs to a Department Group and has set Call Forward-If Busy to C within the Department Group.
• If the system detects Busy signal on B, it forwards the call for B to C in the Department Group.
• However, if the caller has called the Department Group instead of calling B directly, the call will land in the
sequence on all Department group extensions. When it is B's turn, the call will not be forwarded to C, B will
ring instead.
C belongs to a Department Group and has set Call Forward-No Reply to D within the Department Group.
• The system waits for the Call Forward No-Reply Timer to expire, and forwards the call for C to D in the
Department Group.
• Whenever there is a call for D, if the system does not detect a busy signal from D, it waits for the Call
Forward No-Reply timer to expire.
• When there is a call for E, the system rings on both E and the destination F.
Feature Interaction:
• Do Not Disturb (DND): When DND and Call Forward are set on an extension, DND is given priority.
• You can select the types of calls, i.e. internal calls only, or trunk calls, or both, to be forwarded to
external numbers. You can program the system to forward internal calls only, or trunk calls only or both
trunk calls and internal calls, to the external number. For this, the parameter 'Allow External Call
Forward for' must be programmed in the “Station Advanced Feature Template” of the extensions that
are allowed Call Forward.
• The system supports only single-point Call Forward, which means, if the destination extension is also
forwarded, the call will not follow the forwarding path. For example: Calls for extension A are set to be
forwarded to extension B. Call Forward is also set on extension B with C as the destination number.
Calls for A will land on B only and calls for B will land on C only.
• Only one Call Forward Type can be set from an extension. Every new Call Forward Type set overrides
the previous one.
• When the calls are forwarded the extension user gets the feature tone on lifting the handset to indicate
that Call Forward is set on his/her extension.
Call Forward must be enabled in the Class of Service (COS) group of the extensions to which this feature is to be
allowed.
When Call Forward No-Reply is set, if required the Call Forward No-Reply Timer needs to be programmed.
You may select the types of calls, i.e. internal, external, both internal and external calls to be forwarded by
programming the ‘Allow External Call Forward for’ parameter.
If you want to deny Call Forward to certain extensions, follow these steps:
b. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
c. Assign this new Template to the extensions to which Call Forward is to be denied.
Refer the topics Class of Service and Station Basic Feature Template for programming instructions.
The Call Forward No-Reply Timer is to be programmed in the “Station Advanced Feature Template” applied on the
extensions which are allowed Call Forward in their COS.
If you want to set this timer to the same duration for all extensions, simply set the Call Forward No-Reply Timer in
the default Station Advanced Feature Template 01 which is assigned to all extensions.
If you want to set different Timer duration for different extensions, then prepare separate Station Advanced Feature
Templates with the desired Timer durations and assign different Templates (with different Timer durations) to the
extensions as desired.
You may select from 'Internal Calls', 'Trunk Calls' and 'Internal + Trunk Calls'. By default, only trunk calls are
forwarded to external numbers in the default Station Advanced Feature Template 01 which is assigned to all
extensions.
Refer the topic “Station Advanced Feature Template” for instructions on customizing the template and applying the
template to extensions using Jeeves and from a Telephone.
• Click the 'Station Advanced Feature Template' link to open the page.
• Select an Advanced Feature Template number. (by default Template 01 is assigned to all extensions)
• Apply the Template now configured with the Call Forward No-Reply Timer to the extensions.
• Dial command 5602-1-Station Advanced Feature Template Number-02-Call Forward No-Reply Timer
Where,
Station Advanced Feature Template is from 01 to 50. Default: 50.
Timer is from 001 to 255 seconds.
02 is the parameter number for "Call Forward No-Reply Timer" in the Template.
For example: To program Call Forward No-Reply Timer as '60' secs.' in Template number 02, dial 5602-1-
02-02-060
• Exit SE mode.
Refer the topic “Station Advanced Feature Template” for instructions on applying the template to extensions
using Jeeves and from a Telephone.
When Call Forward No-Reply is set on a phone that is programmed in a Trunk Landing Group, the calls will
be forwarded on expiry of 'Call Forward No-Reply' programmed in the routing group for this member
phone. Call Forward No-Reply Timer, programmed in Station Advanced Feature Template will not be
applied in this case.
How to use
Call Forward can be set/canceled by extension users who are allowed this feature. It can be set/canceled by an
extension user for another extension (refer “Call Forward-Remote” to know more).
OR
• Dial 13.
• Scroll to select the desired Call Forward Type.
• Press 'Enter' key.
• Enter destination Phone Number/Voice Mail Group Number/ Department Group Number.
• You get a confirmatory text message and confirmation tone.
• Go Idle or you get dial tone after 3 seconds.
• If the call is to be forwarded to an external number, dial Trunk Access Code, then the external phone
number and terminate the command with #*.
• For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63,
64.
• For users in USA, TAC for dialing external numbers are: 9, 5, 81, 82, 83, 84.
• If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail group. The default
Access Code is 3931. Verify with the System Engineer if the default VMS Group Access Code has
been changed and use the new code to dial the VMS Group.
OR
• Dial 13.
• Select 'Cancel'.
• You get a confirmatory text message and confirmation tone.
• Replace Handset on the cradle or you get dial tone after 3 seconds.
What's this?
An extension user can set Call Forward for another ('remote') extension from his/her own extension. Thus, Call
Forward set for an extension from another extension is called 'Call Forward-Remote'.
This feature can be used by the Operator or the Receptionist to forward the calls for the Managers and other
extension users to the destinations where they will be available.
This feature is also useful in Hotels, where the Front Desk can set Call Forward for guests. Refer the ETERNITY
Hospitality System Manual to know how this feature can be used in hotels.
• Call Forward-Remote is possible only from the System Administration (SA) mode.
How it works
This feature works in the same way as Call Forward. The only difference is that it is set by one extension user for
another extension.
For example:
• A needs to forward calls for B's extension to another extension 'C' or an external number or a Voice Mail
Group or a Department Number.
• A dials the Call Forward-Remote feature code followed by B's extension number, the destination number
where the calls for B should land.
• The system routes all incoming calls for B to the destination number.
How to configure
As Call Forward-Remote can be invoked only from the SA mode, either the feature 'System Admin Mode Access' or
'System Admin Station' must be enabled in the Class of Service of extensions that are to be allowed this feature.
The feature 'System Admin. Mode Access' requires a password to be dialed. Users must be provided a
password to use this feature from their extensions. The feature 'System Admin Station' allows entry into SA
mode, without a password.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. This Template is assigned CoS group 01 by default. The default CoS group 01 has only 'System Admin
Mode Access' enabled. So, all extensions of the ETERNITY can access Call Forward Remote, provided they have
the SA password.
You may decide which extensions should be allowed Call Forward-Remote feature. In general practice only very
few extensions are allowed this feature.
a. Define a CoS group with either 'System Admin Mode Access' or 'System Admin Station' enabled. Recall
that the facility 'System Admin Mode Access' is password protected, so the extensions allowed access to
this feature must also be provided an SA Password.
b. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
c. Assign this new Template to the extensions to which Call Forward-Remote is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
How to use
OR
• Dial 1072-006.
• Enter the Destination Phone Number.
• Scroll to select the desired Call Forward Type:
• All Calls.
• If Busy.
• If No Reply.
• If Busy or No Reply.
• Dual Ring.
• Press 'Enter' key.
• Enter Destination Phone Number135/Voice Mail Group Number136/Department Group.
• You get a confirmation tone and a text message for the Call Forward type set.
• Go Idle or you get dial tone after 3 seconds.
135. If call is to be forwarded to an extension of the ETERNITY, dial the extension number. If call is to be forwarded on an external num-
ber, dial Trunk Access Code, then dial the external phone number and terminate the command with #*.
For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63, 64. For users in USA, TAC for
dialing external numbers are: 9, 5, 81, 82, 83, 84.
136. If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail group. The default Access Code is 3931.
OR
• Dial 1072-006.
• Enter Extension Number.
• Scroll to select 'Cancel'.
• Press 'Enter' key.
• You get a confirmation tone and text message for Call Forward canceled.
• Go Idle or you get dial tone after 3 seconds.
• Lift handset.
• Dial 1072-006.
• Enter Extension Number.
• Dial 1 for All Calls
• Dial 2 for If Busy
• Dial 3 for If No Reply
• Dial 4 for If Busy or No Reply
• Dial 5 for Dual Ring
• Dial destination Phone Number/Voice Mail Group Number.
• You get confirmation tone.
• Replace handset.
What's this?
Extension users may want their calls to be automatically forwarded to a desired destination number during working
hours or non-working hours. To cite an example, a Support Technician spends working hours on the field and
wants all incoming calls on his extension in the office to be forwarded to his cell phone during working hours. During
non-working hours, he wants call calls to be forwarded to his voice mail.
Remembering to set and cancel Call Forward and changing the destination number for each Time Zone, i.e.
working hours, non-working hours, break hours, every day proves to be cumbersome for such extension users.
In addition to “Call Forward”, ETERNITY supports 'Call Forward - Scheduled', which allows extension users to set
call forward for desired Time Zones at one time, and the system automatically forwards the calls to the destination
defined for each Time Zone.
How it works
Call Forward-Scheduled supports all the forwarding options as Call Forward: Unconditionally, If Busy, If No Reply, If
Busy or No Reply, Dual Ring.
Any of these options can be set for the three Time Zones: working hours, break hours and non-working hours.
The destination for Call Forward-Scheduled can be an internal (extension) number or an external number.
Both 'Call Forward' and Call Forward-Scheduled can be set on the same extension. In this case, priority is given to
'Call Forward' over Call Forward-Scheduled.
The logic for forwarding calls to the destination number remains the same as described in the topic “Call Forward”,
illustrated in the following example.
• When there is a call on extension A, the system first checks if there is any 'Call Forward' type (i.e.,
Unconditional, Busy, No Reply, Busy/No Reply, Call Follow Me) set on extension A.
• If 'Call Forward' is set on extension A, the system will follow the logic described in 'How it works' under the
topic 'Call Forward".
• If no 'Call Forward' is set on extension A, the system will check if Call Forward-Scheduled is set on A.
• Since Call Forward-Scheduled is set on extension A, the system will compare the Time Zone for which the
Call Forward is scheduled with the current Time Zone of extension A.
• If the current Time Zone of extension A is the same as the Time Zone set for Call Forward Scheduled, i.e.
non-working hours, the call will be forwarded to extension B as per the call forward type set.
• As the Call Forward Type set by A is Unconditional, the system will forward the call to B, without checking
for the Busy Tone and without waiting for the Call Forward No-Reply Timer to expire.
• Call Forward - Scheduled can be set simultaneously for more than one Time Zone from the same
extension. For example, extension A can set Call Forward-Scheduled for working hours, then again set
Call Forward-Scheduled for non-working hours, and again for break hours.
• A different Call Forward Type can be set for a different Time Zone. For example, extension A can set
Call Forward -Unconditional for non-working hours, and Call Forward -Busy for working hours. Also, a
different destination number can be set for forwarding calls in each Time Zone. For example, extension
A can set Call Forward-Unconditional for non-working hours to a mobile number and set extension B as
destination number for working hours.
• When more than one Call Forward type is set on the same extension for the same Time Zone, the
latest Call Forward type set for the Time Zone will override the previous Call Forward type set for that
Time Zone. For example, extension A sets Call Forward -Busy for working hours, then sets Call
Forward Busy or No Reply for working hours, the latter will override the former. The system will
consider the latest, i.e. Busy or No Reply as the Call Forward type for forwarding calls during working
hours.
• Call Forward-Scheduled can be cancelled individually for a desired Time Zone or all at once for all
Time Zones.
• Call Forward-Scheduled can be set by extension users as well as for extension users from the System
Administrator mode.
• It is also possible to select the types of calls, i.e. internal calls only, or trunk calls, or both, to be
forwarded to external numbers. You can program the system to forward internal calls only, or trunk
calls only or both trunk calls and internal calls to the external number. For this, the parameter 'Allow
External Call Forward for' must be programmed in the “Station Advanced Feature Template” of the
extensions that want to use Call Forward-Scheduled.
How to configure
The programming of this feature involves the same parameters as in “Call Forward”.
'Call Forward' must be enabled in the Class of Service (CoS) group of the extensions to which this feature is to be
allowed. Refer the topic "Call Forward".
If Call Forward No-Reply is to be set, and if required, the Call Forward No-Reply Timer may be
Programmed in the “Station Advanced Feature Template” applied on the extensions which are to be allowed this
feature. Refer the topic “Call Forward”.
The types of calls to be forwarded to the external number may be selected in the parameter "Allow External Call
Forward for" in the “Station Advanced Feature Template” applied on the extensions which are allowed Call
Forward-Scheduled. You may select from 'Internal Calls', 'Trunk Calls' and 'Internal + Trunk Calls'. By default, only
trunk calls are forwarded to external numbers.
Extensions that are to be allowed to set Call Forward-Scheduled for other extensions must be allowed either the
feature 'System Admin Mode Access' or 'System Admin Station' in their COS. Refer the topic “Call Forward-
Remote”.
• The destination number for forwarding calls can be a maximum of 24 digits. Terminate the command
with #* if destination number has fewer than 24 digits.
• If the destination number is an external number, enter the Trunk Access Code followed by the
destination number.
OR
• Dial 1175.
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to the desired Call Forward type for the selected Time Zone.
• Press Enter key to select Call Forward type.
• Enter Destination Number on prompt.
• You get confirmation tone and message showing extension to which Call Forward is set.
OR
• Dial 1175.
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to select Cancel.
• Press Enter key.
• You get confirmation tone and message.
OR
• Dial 1175.
• Scroll to 'Cancel Call Forward'.
• Press Enter key.
• You get confirmation tone and message.
OR
• Dial 1072-223.
• Enter extension number (from which calls are to be forwarded)
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to the desired Call Forward type for the selected Time Zone.
• Press Enter key to select Call Forward type.
• Enter Destination Number on prompt.
• You get confirmation tone and message showing extension to which Call Forward is set.
OR
• Dial 1072-223
• Enter extension number (for which it is to be canceled)
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to select Cancel.
• Press Enter key.
• You get confirmation tone and message.
OR
• Dial 1072-223
• Enter extension number (for which it is to be canceled)
• Scroll to 'Cancel Call Forward'.
• Press Enter key.
• You get confirmation tone and message.
• Lift handset.
• Dial 1175-1-1-Destination Number for CF-Scheduled-Unconditional.
• Dial 1175-1-2-Destination Number for CF-Scheduled -Busy.
• Dial 1175-1-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1175-1-4-Destination Number for CF-Scheduled-Busy/No Reply.
• Lift handset.
• Dial 1175-2-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1175-2-2-Destination Number for CF-Scheduled -Busy.
• Dial 1175-2-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1175-2-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1175-2-5-1 for CF-Scheduled -Dual Ring.
• Dial 1175-2-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1175-2-0 to cancel CF-Scheduled for Break Hours.
• Replace handset.
• Lift handset.
• Dial 1175-3-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1175-3-2-Destination Number for CF-Scheduled -Busy.
• Dial 1175-3-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1175-3-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1175-3-5-1 for CF-Scheduled -Dual Ring.
• Dial 1175-3-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1175-3-0 to cancel CF-Scheduled for Non-working Hours.
• Replace handset.
• Lift handset.
• Dial 1072-223-Extension number-1-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1072-223-Extension number-1-2-Destination Number for CF-Scheduled -Busy.
• Dial 1072-223-Extension number-1-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1072-223-Extension number-1-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1072-223-Extension number-1-5-1 for CF-Scheduled -Dual Ring.
• Dial 1072-223-Extension number-1-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1072-223-Extension number-1-0 to cancel CF-Scheduled -for working hours.
• Replace handset.
• Lift handset.
• Dial 1072-223-Extension number-2-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1072-223-Extension number-2-2-Destination Number for CF-Scheduled -Busy.
• Dial 1072-223-Extension number-2-3-Destination Number for CF-Scheduled -No Reply.
• Lift handset.
• Dial 1072-223-Extension number-3-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1072-223-Extension number-3-2-Destination Number for CF-Scheduled -Busy.
• Dial 1072-223-Extension number-3-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1072-223-Extension number-3-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1072-223-Extension number-3-5-1 for CF-Scheduled-Dual Ring.
• Dial 1072-223-Extension number-3-5-0 to cancel CF-Scheduled-Dual Ring.
• Dial 1072-223-Extension number-3-0 to cancel CF-Scheduled - for break hours.
• Replace handset.
What is this?
Call Park allows you to place a call on hold, so it can be retrieved from the same or another extension of the
system.
A call is 'parked' when the extension user temporarily places the call into a location in the system called 'Orbit'. The
user can attend to other calls. The parked call can be retrieved on completion of the current call by dialing the Orbit
number.
Call Parking is useful in offices housed in different parts of a building or multi-storied offices. It is useful in situations
like:
• the person who picked up the call is not the desired called party or the desired party is at an unknown
location. The person who picked up the call can then either go to find the desired called party or call other
numbers to find him/her. When found, the desired called party can pick up the call from the same or any
extension by dialing the Orbit number.
• the person who picked up the call may have to go to another part of the office to look up a file or consult a
colleague. The person can park the call and continue the conversation from the other part of the office.
• Call Park-General Orbit: The extension user can park calls in any of the 8 'general' Orbits, which are like
fictional extensions located in the system. The calls parked in the General Orbit can be picked up from any
extension by dialing the General Orbit Number. At a time, only one call can be parked in each General
Orbit.
• Call Park-Personal Orbit: Each telephone instrument (EON/SLT/SIP extension) connected as extension
has one Personal Orbit. Calls parked in personal orbit can be picked up only from where the call is parked.
So, no other person can pick up this call. Multiple calls can be parked in the Personal Orbit at a time.
Extension users can park the call either in the General Orbit or the Personal Orbit by dialing an Orbit Number from
1 to 9, where:
After parking a call, the extension user can continue to make and answer other calls and use other system features.
However,
• If neither A nor B retrieves the parked call within the Call Park Timer, the system will hunt for the extension
that parked the call (A) on the expiry of the Call Park Timer.
• Meanwhile, if A is busy, the system again keeps the call parked in orbit number 2 for the period of the Call
Park Timer. This process continues for the duration of the Call Park Release Timer, which is set to 3
minutes by default.
• If A is free, the system will ring on A's phone. A gets connected to C again.
• If A does not retrieve the parked call till the end of the Call Park Release Timer, C gets disconnected.
When there are multiple calls to be retrieved from the Personal Orbit, they are retrieved one by one, without
following any particular sequence like FIFO or LIFO.
To be able to use 'Call Park', this feature must be enabled in the COS of the requesting extension.
However, for retrieving parked calls, the system does not check COS. So any extension can retrieve
parked calls.
How to configure
• If required, you may change the duration of the Call Park Timer and the Call Park Release timer. See
“System Timers and Counts” for instructions.
To retrieve a parked call from your phone, when your phone is in idle state:
To retrieve a parked call from your phone, when you are in speech with someone:
• Press Hold.
• Press DSS Key assigned to 'Call Park - Retrieve'.
• Dial 116
OR
• Enter Orbit Number where you parked the call (1-9)
(Personal Orbit:1; General: 2-9).
To retrieve a parked call from your phone, when your phone is in idle state:
To retrieve a parked call from your phone, when you are in speech with someone:
What's this?
Call Hold enables you to put an on-going conversation (with an internal or external number) on hold, and call
another person or receive a call from another person. You can retrieve the call you put on hold, after the
conversation with the other party has ended or in the middle of the conversation with the other party.
Call Hold is a feature of the DKP. ETERNITY enables two types of Call Hold on the DKP: Exclusive Hold and Global
Hold.
Exclusive Hold
The call placed on Exclusive hold can be retrieved only from the DKP which put it on hold. The call remains
connected to the DKP which placed it on hold.
When a call is put on Exclusive hold, the ETERNITY starts the Call Park Timer and Call Park Release Timer.
The call remains on hold for the duration of the Call Park Timer (programmable; default: 45 seconds). If this call is
not retrieved before the Call Park Timer expires, the call is parked in the Personal Orbit for the duration of the Call
Park Release Timer (programmable; default: 3 minutes). If the DKP becomes idle within this Timer, the call is
returned to the DKP and the DKP gets Ring Back Tone.
If the DKP is busy, the system waits for the duration of the Call Park Release Timer for the DKP to go idle or
retrieve the call. If the DKP goes idle within this Timer, the call is returned to the DKP and the DKP gets Ring Back
Tone.
The call is disconnected if the DKP is still not free or does not retrieve the call before the Call Park Release Timer
expires.
• Pressing the Key of the Call Appearance of the call put on hold (before it gets parked).
• Answering the call, when it returns at the end of the Call Park Timer.
Global Hold
The call placed on Global hold can be picked up from any DKP extension of ETERNITY. The call remains
connected in the system. The call remains on hold for the duration of the Call Hold Retrieval Timer (programmable;
default: 60 seconds). If this call is not retrieved before the expiry of the Timer, the call is returned to the DKP which
put it on hold.
To be able to place calls on Global Hold, you must enable 'Global Hold' in the System Parameters of ETERNITY.
The DKP (which picks up the call) must have a DSS Key to access the Trunk or the Extension which is put on hold.
• Pressing the DSS key assigned to the extension put on Global Hold.
The call on Global Hold must be picked up before the Call Hold Retrieval Timer expires.
ETERNITY provides the flexibility to use Exclusive Hold and Global Hold at the same time. You can put
calls on Exclusive Hold even when Global Hold is enabled in the system.
How to configure
For this feature to work, you must program the following parameters:
• Class of Service: Call Hold must be enabled in the Class of Service (CoS) of the DKPs you want to allow
this feature.
In the default Station Basic Feature Template 01 assigned to all extensions of ETERNITY, Call Hold is
included in the 'Basic Features' assigned to all Class of Service groups, including the default CoS group
01. So, all extensions of ETERNITY can use this feature.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” to know more.
• Global Hold: Enable this flag in the System Parameters if you want Calls on Hold to be picked up by any
DKP extension.
• DSS Keys: Program DSS Keys for Trunks and Stations on the DKPs which are allowed to retrieve calls on
Global Hold. Refer the topic “DSS Keys Programming” for instructions.
• Call Hold Retrieval Timer: Change the default setting this timer to the desired duration, if required.
• Call Park Timer: Change the default setting this timer to the desired duration, if required.
• Call Park Release Timer: Change the default setting this timer to the desired duration, if required.
• Scroll further to 'Call Hold Retrieval Timer'. Set the Timer to the desired duration.
• Go to 'Call Park Release Timer'. Set the Timer to the desired duration.
• Exit SE mode.
How to use
Exclusive Hold
To put a call on Exclusive Hold, when Global Hold is disabled:
Global Hold
To put a call on Global Hold:
What's this?
ETERNITY stores the details of 20 each, of the following types of calls:
• Missed calls: incoming calls that were not answered by extension users.
• Answered calls: incoming calls answered by extension users.
• Dialed calls: calls made by extension users.
The call history of each of the above types of calls is stored by Name, Number, and Date-Time of the Call.
If there is no name in the CLI of the above types of calls, the system stores and displays the Number and the Date-
Time. In case there is no number in the CLI, the system will display the Port number on/from which the call was
received/made.
The Call Logs contain details of both internal as well as external calls made or received by the extension users.
• view call history: you can see the calls you missed, answered or dialed.
• make calls: you can call any number that you have missed, answered, or dialed.
• edit the numbers: you can change or modify the number in the call log. This is useful when the CLI
received and stored in the call log is not in the same format that is to be used to make calls.
• save the numbers: you can store the external numbers in your call logs in the "Personal Directory" and
use them for “Personal Abbreviated Dialing”.
The maximum number of calls that can be stored under each Call Log type is 20. The logs will be cleared
automatically using the First-In, First-Out method, i.e. the latest call detail will replace the record of the oldest call
detail.
Given the limited Call Log capacity, the system also allows you to choose if you want internal calls to be displayed
or not in the Missed, Answered and Dialed Call Logs. And accordingly it will store internal calls in the logs.
The system stores each Missed, Answered and Dialed call individually even if the same number is received
multiple times.
How to configure
This feature does not require any specific programming, except:
• Selecting whether internal calls should be logged in the Missed, Answered and Dialed Call Logs. This can
be done on the 'System Parameters' page of Jeeves or by using a Telephone.
• Programming of a DSS key for the Call Logs feature. For instructions please refer the topic “Configuring
DKP Extensions”.
• Exit SE mode.
How to use
The Call Logs feature allows you to view calls and edit numbers, make calls to any number logged, and store
numbers.
• If there is no name in the CLI, the Call Log will only display the number.
• If you press the 'Enter' key, the system will dial out the number you just viewed.
OR
• Press the DSS Key assigned the Call Logs feature, when it glows.
• The phone will display the call log details of the last missed call by: <Name> <Date> <HH:MM> (only if
name is received).
• Press Enter key.
• The phone will display the Number: <XXXXXXXXXXX>
• You may exit the Phone Menu by going OFF-Hook or pressing the Cancel key138.
• You may also edit or store the number.
• You may scroll with the < Back navigation key to view the other call logs.
• The LED of the Call Logs DSS key will be turned off once you have viewed the missed call.
137. This key is available on EON48. This key must be programmed on EON42.
138. This key is available on EON48. This key must be programmed on EON42.
The original number (you now changed) will remain unaffected in the Log. However, if you make a call to
the new number (you changed), it will be logged in the "Dialed" call log and the Last Number Redial list.
• When you store the number in the Personal Directory, the system will automatically assign Trunk
Access Code "TAC-1".
• If all 25 Location Index Numbers of the Personal Directory are already programmed, the message
"Memory Full" will appear on your phone's display and you will get an Error Tone. Refer the topic
“Abbreviated Dialing” to know more.
What's this?
Call Pick-Up allows extension users to answer calls ringing on other extensions from their own extension; without
physically going to the ringing extensions.
Extension users can 'pick-up' both internal and trunk calls ringing on other extensions.
As extension users can answer calls of their colleagues or co-workers without physically going to their extensions,
this feature ensures that all incoming calls are answered.
• Call Pick Up-Group - extensions are assigned to Pick-Up Groups. Any extension in a Pick-Up Group can
answer calls ringing on other extensions within the same group only.
• Call Pick-Up Selective - calls ringing on any extension of the system can be answered.
How it works
Call Pick-Up Group
• Extensions must be assigned to Call Pick-Up Groups. The extensions in a Call Pick-Up group may be SLT,
DKP and ISDN Terminal.
• For example, extensions 2007, 2008, 2009, 2010. 2011, 2012, 2013 are assigned to Pick-Up Group
number 03.
• When an extension in this group rings, any extension in the group can pick up the call by dialing the
feature access code for "Call Pick-Up Group" (default: 4).
• Whenever an extension in the system rings, the call can be picked up by any extension of the system by
dialing the feature access code and the number of ringing extension.
When more than one extension in a Pick-Up Group is ringing, you can choose which one to answer first,
using Call Pick-Up Selective.
• Call States: Call Pick-Up will fail if the ringing extension goes into idle state just when you are dialing the
pick-up access code.
• Auto Call Back: Call Pick-Up will fail if the call ringing on the extension is an Auto Call Back request.
• Alarms: Call Pick-Up will fail if the call ringing on the extension is an Alarm Call.
How to configure
For this feature to function, 'Call Pick-Up' should be enabled in the Class of Service of extension that are to be
allowed this feature.
On a sheet of paper, list the extensions that are to be grouped into a Call Pick-Up Group. Make as many Call Pick-
Up Groups as required. Assign each group a number.
Call Pick-Up
SLT Extensions DKP Extensions ISDN Terminals
Group Number
01 2002, 2003, 2006, 2014 3003, 3004, 3005, 3201, 3201, 3203
99
The numbering of Call Pick-Up Groups must start from 01 and end at 99.
Do not assign '00' as Call Pick-Up Group. '00' is the command to de-assign from a Call Pick-Up Group.
To program these groups, you may use Jeeves or issuing SE commands from a telephone.
• In the column Call Pick-up Group, assign the group number for SLT extensions. Refer to the sheet of paper
you prepared.
• Assign Call Pick-Up Group number to DKP extensions. Refer to the sheet of paper you prepared.
• Assign Call Pick-Up Group numbers to ISDN Terminals. Refer to the sheet of paper you prepared.
• Exit SE mode.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. The Station Basic Feature Template 01 is assigned COS group 01 which has Call Pick-Up feature
enabled. Thus, all the extensions of the ETERNITY can use Call Pick-Up by default.
If you want to deny Call Pick-up feature to all extensions, you can simply disable Call Pick-Up in the default CoS
group 01.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the extensions to which Call Pick-Up is to be denied.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
How to use
OR
• Dial 4.
• Talk.
• Go idle.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
OR
• Dial 12.
• Dial number of the Extension you want to pick up.
• Talk.
• Go idle.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
What's this?
Call Progress Tones (CPT) are audible tones sent from switching systems such as PSTN or PBX to calling parties
to show the status of phone calls, like dial tone, error tone, ringing error in number dialed, ringing called party, busy
line, etc.
Each CPT has a distinctive tone frequency and cadence assigned to it, for which some standards have been
established by the International Telecommunication Union (ITU).
On the basis of specific frequency, modulating frequency and cadence, the CPTs generated by ETERNITY are
categorized as:
Dial Tone 1 Played on lifting the Toooooooooooo Played for 7 seconds. Dial Tone Timer
handset. After which Error Tone
starts
Dial Tone 2 Played on lifting the Toooooooooooo Played for 7 seconds. Dial Tone Timer
handset, when 'Store After which Error Tone
and Forward Dialinga' is starts
done.
Ring Back Tone Played when the Turroo... Turrroo Played for 45 seconds Ring Back Tone
internal number you Timer
have dialed is free.
Busy Tone High pitch beeps with Tooooooo......... Played for 7 seconds. Busy Tone Timer
(Engaged Tone) equal ON and OFF Toooooooo
periods, played when
the dialed extension is
busy. Busy tone
continues for 7
seconds. This Busy
Tone Timer is
programmable.
Error Tone Fast beeps, played on a Too…Too…Too Played for 30 seconds Error Tone Timer
(Congestion/ wrong operation being …Too
Refusal Tone as performed or a feature
per ITU invoked without access.
Internal Call Short beep followed by Beep.……….… Played for duration of Interrupt Request
Waiting Tone longer OFF duration Beep the Interrupt Request Timer, Barge-In
(Intrusion Tone repeated every second; Timer or the Barge-In Timer
as per ITU) played to the busy Timer.
extension when another
extension attempts
Interrupt Request/
Barge-In
External Call Two ticks followed by a Beep...Beep… Played for the duration Transfer-On Busy
Waiting Tone longer OFF time of ……......Beep... of the Transfer-On Busy Timer.
(Call Waiting approx. 3 seconds; Beep Timer.
Tone as per ITU) played to a busy
extension when there is
a new incoming Trunk
call.
Confirmation Continuous, fast beeps, Beep... Beep... Played for 7 seconds. Confirmation Tone
Tone played to confirm Beep Timer
(Acceptance successful use of
Tone as per ITU) features.
Feature Tone Short beep followed by Beep................. Played until user goes
a longer off duration Beep ON-Hook or dials a
repeated every second; feature code.
played when dialing
feature access codes
Programming Short beep followed by Beep................. Played until user goes Programming Tone
Tone a longer off duration Beep ON-Hook or dials a Timer
repeated every second; command.
played to prompt
entering of fresh
commands during
programming.
Programming Continuous, fast beeps; Beep... Beep... Played for 3 seconds. Programming
Confirmation played to indicate that Beep Confirmation Timer
Tone system has received a
valid command and is
processing it.
Programming Fast beeps, played on a Too…Too…Too Played for 3 seconds. Programming Error
Error Tone wrong programming …Too Tone Timer
command being dialed.
a. In Store and Forward dialing, the digits are first stored in a memory location and then these are dialed on the trunk.
For example: When Least Cost Routing (LCR) is enabled, the system will store the dialed digits first, check the trunk
through which the call is to be routed and then dials the number on the appropriate trunk.
Tone standards vary with the country of application. For example, as per ITU standard, the Dial Tone for India
consists of 400Hz modulated by 25Hz, whereas it is 350+440Hz, without modulation, for USA/Canada. Further,
many countries use different frequencies and cadences for the same tone. For example, in the US, five different
frequency and cadence are used for Dial Tone.
ETERNITY offers the flexibility of setting the Call Progress Tone Generation (CPTG) type to match the country-
specific CPT standards established by ITU.
India being the default 'Region' for ETERNITY, the CTPG for India is set as default in the system.
For countries that use different frequencies and cadences for the same tone, for instance, USA, only one
frequency/cadence among the group is considered. See Table "Default CPTG Type".
How to configure
Programming of Call Progress Tones involves configuration of three parameters: CPTG Type (Region), CPT
related Timers, and Dial Tone Type.
The country-specific CPTG type is set automatically by the system when the 'Region' is selected. However, if
required, the System Engineer can change the CTPG type set by the system.
• Click 'Call Progress Tones' under Regional Settings to open the page.
CPTG Dial tone 1 Dial Tone 2 Ring Back Tone Busy Tone
Region Region Cadence Cadence Cadence Cadence
Code Freq. Freq. Freq. Freq.
(sec) (sec) (sec) (sec)
1 Region1 440 Continuous 350+440 Continuous 350+440 0.4on 0.2off 440 0.75on 0.75off
0.4on 2.0off
2 Region2 400 Continuous 400 Continuous 400 0.6on 0.2off 400 0.5on 0.5off
0.2on 2.0off
3 Region3 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
4 Argentina 425 Continuous 425 Continuous 425 1.0on 4.0 off 425 0.3on 0.2off
5 Australia 425*25 Continuous 425*25 Continuous 400*25 .4on .2off 425 0.375on
.4on 2.0off 0.375off
6 Brazil 425 Continuous 425 Continuous 425 1.0on 4.0 off 425 0.25on 0.25off
7 Canada 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
8 China 450 Continuous 450 Continuous 450 1.0on 4.0off 450 0.35 on
0.36off
9 Egypt 425*50 Continuous 425*50 Continuous 425*50 2.0on 1.0off 425*50 1.0on 4.0off
10 France 440 Continuous 440 Continuous 440 1.5on 3.5off 440 0.5on 0.5off
11 Germany 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.48on 0.48off
12 Greece 425 0.2on 0.3off 425 0.2on 0.3off 425 1.0on 4.0off 425 0.3on 0.3off
0.7on 0.8off 0.7on 0.8off
13 India1 400*25 Continuous 400*25 Continuous 400*25 .4on .2off 400 0.75on 0.75off
.4on 2.0off
14 Indonesia 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
15 Iran 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
16 Iraq 400 0.4on 0.2off 400 0.4on 0.2off 400 Continuous 400 1.0on 1.0off
0.4on 1.5off 0.4on 1.5off
17 Israel 400 Continuous 400 Continuous 400 1.0on 3.0off 400 0.5on 0.5off
18 Italy1 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
19 Japan 400 Continuous 400 Continuous 400*25 1.0on 2.0off 400 .5on .5off
20 Kenya 425 Continuous 425 Continuous 425 0.67on 425 0.2on 0.6off
3.0off 1.5on 0.2on 0.6off
5.0off
21 Korea 350+440 Continuous 350+440 Continuous 440+480 1.0on 2.0off 480+620 0.5on 0.5off
22 Malaysia 425 Continuous 425 Continuous 425 0.4on 0.2off 425 0.5on 0.5off
0.4on 2.0off
23 Mexico 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.25on 0.25off
24 New 400 Continuous 400 Continuous 400+450 0.4on 0.2off 400 0.5on 0.5off
Zealand 0.4on 2.0off
25 Phillippines 425 Continuous 425 Continuous 425+480 1.0on 4.0off 480+620 0.5on 0.5off
26 Poland 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
27 Portugal 425 Continuous 425 Continuous 425 1.0on 5.0off 425 0.5on 0.5off
28 Russia 425 Continuous 425 Continuous 425 0.8on 3.2off 425 0.4on 0.4off
29 Saudi 425 Continuous 425 Continuous 425 1.2on 4.6off 425 0.5on 0.5off
Arabia
30 Singapore 425 Continuous 425 Continuous 425*24 0.4on 0.2off 425 .75on .75off
0.4on 2.0off
31 South 400*33 Continuous 400*33 Continuous 400*33 0.4on 0.2off 400 .5on .5off
Africa 0.4on 2.0off
32 Spain 425 Continuous 425 Continuous 425 1.5on 3.0off 425 0.2on 0.2off
33 Thailand 400*50 Continuous 400*50 Continuous 400 1.0on 4.0off 400 0.5on 0.5off
34 Turkey 450 Continuous 450 Continuous 450 2.0on 4.0off 450 0.5on 0.5off
35 UAE 350+440 Continuous 350+440 Continuous 400+450 0.4on 0.2off 400 0.375on
0.4on 2.0off 0.375off
• Exit SE mode.
For SE commands to change Interrupt Request Timer and the Barge-In Timer, Transfer-On Busy Timer,
refer the relevant topics: “Interrupt Request (IR)”, “Barge-In” and “Call Transfer”.
How to use
It is important that users of ETERNITY also get acquainted with the different Call Progress Tones played by the
system, so that they understand the meaning of the terms used for various tones. Therefore, ETERNITY makes it
possible for users to listen to the various Call Progress Tones.
Demonstration of Tones
It is possible to demonstrate Call Progress Tones to users by dialing the SE commands from EON or an SLT.
By default, the system will play each tone as demonstration for 30 seconds. The duration of demonstration can be
changed by setting the 'Tone Demo Timer' to match user preference (see "Changing CPT-related Timers using a
Telephone" above).
140. Time for which the system demonstrates the tone/ring to the user.
• Exit SE mode.
What's this?
When the VoIP Card of ETERNITY is connected to a public IP network, it is may be necessary to restrict traffic to
and from a particular IP address only.
With the feature 'Call Restriction based on IP Address', ETERNITY makes it possible to entertain requests on its
VoIP Ethernet ports for predefined IP Addresses only,
How it works
For this feature to work,
• the "IP Address based call traffic restriction" flag must be enabled for the VoIP Ethernet Port, and
• the "White List IP Address Table", i.e. a list of IP Addresses and their respective Subnet Masks from where
the traffic is to be allowed, must be configured for VoIP Ethernet Port.
• With flag enabled and the table programmed, traffic coming from all IP Addresses, other than those
programmed in the White List, will be blocked.
• If the flag "Call Restriction based on IP Address" is enabled, but the White List IP Address Table is
blank, all incoming traffic will be rejected and it will not be possible to make calls on such a VoIP Card.
• Call Restriction will be applied also on all the SIP Trunks which are assigned to the VoIP Ethernet Port.
How to configure
• Decide on which of the VoIP Ethernet Ports IP Address based call traffic restriction is to be applied.
• For each VoIP Ethernet Port you want to create a White List Table, make a three column table on a sheet
of paper.
• Make a list of IP Addresses. You are allowed to program a maximum of 5 IP Addresses. For each IP
Address enter the corresponding Subnet Mask address.
• With the White List Tables ready, you may program the tables using Jeeves or a telephone.
• Exit SE mode.
What's this?
Call Taping allows extension users to record the telephone conversations they have with other extensions or
external numbers, without the opposite party coming to know about it.
Feature is useful for keeping records of important conversations. For this feature to work, the system must have
either a VMS Card installed in it or be interfaced with Matrix CadencePro.
Calls are taped in a common mailbox assigned to this feature. Extension users with access to the mailbox can
retrieve and listen to the recorded conversations.
To be able to record external incoming and outgoing calls, a list of phone numbers (both incoming and outgoing)
must have been programmed in the system.
Incoming calls without Calling Line Identification (CLI) can also be taped. For this, the flag 'Tape Calls Without CLI?'
must be enabled in the Station Advanced Feature Template.
To be able to record internal calls, the 'Call Taping for Internal Calls flag' must be enabled on the extension which
desires to use this feature.
• Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
How it works
• A and B are extensions. Both have Call Taping parameters configured.
A calls C
• The system matches the dialed number with the numbers in the Number List - Outgoing Calls.
• The system finds a match. When speech is established, the system starts recording the conversation
between A and C automatically in E's mailbox.
• The system matches the incoming number with the numbers in the Number List-Incoming Calls.
• On finding a match, system records the speech between D and B in E's mailbox.
• Call Taping Beeps will be played to D and B only if this feature is enabled.
• If an incoming call does not have any CLI, the system checks the flag 'Tape Calls without CLI' in the
Call Taping parameters.
A calls B
• If the flag is enabled, the system records the speech between A and B in E's mailbox.
• Call Taping Beeps will be played to A and B only if this feature is enabled.
• If the flag is disabled, the speech between A and B will not be recorded.
• The same is done when B calls A. The speech will be recorded in E's mailbox.
Feature Interaction:
• Conversation Recording: If Call Taping and “Conversation Recording” both are enabled for an
extension, then priority is given to Conversation Recording.
How to configure
The functioning of this feature requires the following parameters to be programmed:
• Mailbox Port: You must program the software port of the extension in whose mailbox the calls are to be
taped.
• Taping Calls without CLI Flag: This flag must be enabled if you want calls without CLI to be taped.
• Number Lists for Incoming and Outgoing Calls: The Call Taping Number List-Incoming Calls and Call
Taping Number List-Outgoing Calls are to be programmed so that the system can match the phone
numbers of the incoming and outgoing calls and initiate the recording of the speech.
On a sheet of paper, prepare the Call Taping List Incoming and Call Taping List Outgoing.
001
002
999
Use this table to program the Number lists. By default Number List 09 is assigned for numbers of incoming
calls, and Number List 10 is assigned to numbers of outgoing calls.
• Call Taping Internal Flag: This flag is to be enabled in the Station Advanced Feature Template applied on
those extensions that are to be allowed Call Taping of internal calls, i.e. calls made or received by them to
or from other extensions.
• Call Taping Recording Beeps: This flag is to be enabled if Call Taping Beeps are to be played to the two
parties in speech. Enable Call Taping Beeps only when you want indication of speech recording to the two
parties in speech. By default, this flag is enabled.
• Enter the 'Port No.', i.e. Software Port Number of the SLT/DKP you selected as the Port Type.
• On the same 'System Parameters' page, go to 'Play Beep when Call Taping and Conversation Recording
Starts' and enable/disable beeps by selecting/clearing the check box.
By default, Station Advanced Feature Template 01 is assigned to all extensions of the ETERNITY. If you
want to assign Call Taping facility to all extensions, then program the Call Taping related flags and Number
Lists, in Template 01.
3. Apply this new template to all SLT and DKP stations that are to be allowed this feature.
• Scroll with the horizontal bar to reach the 'Call Taping' column of the Template Number assigned to the
extensions.
• If you want calls without CLI to be taped, click the check box to enable the flag - Tape Calls coming without
CLI?
• To program the list of numbers of incoming calls, click the link Number List- Incoming.
• Click '001-255' link of the default number list 9-10 assigned to Call Taping.
• If the same incoming and outgoing numbers are to be programmed for all extensions, you may simply
program the default Number lists 09 and 10.
• If different incoming and outgoing numbers are to be programmed for different extensions, then
prepare different number lists.
• Enter the List of Incoming Numbers that the system should match in List No. 09.
• Enter the List of Outgoing Numbers that the system should match in List No. 10.
You can program as many as 999 numbers in each list. Each entry on these Lists is stored in a serial order
against a 'Location Index, starting from 001-999'. There are 250 Location Index on each page on your
screen. To go to the next set of Location Index, for instance, 251-500, click the link under 09-10.
• Click 'Submit' at the bottom of the page to save your number lists.
• Follow the same steps to program a different Call Taping number list. But ensure that the different List
number you programmed is entered in the Station Advanced Feature Template applied to the extensions.
• If you want calls between extensions to be taped, click the check box to enable the flag - Call Taping for
Internal Calls.
• Now, apply the programmed template to DKP and SLT extensions to which you want to provide the Call
Taping facility. Refer the topic “Station Advanced Feature Template” for programming instructions.
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
To enable Call Taping Internal Flag in a Station Advanced Feature Template, dial:
• 5602-1-Template Number-Feature Number-Code
Where,
Template Number is from 01 to 50.
Feature Number for Call Taping Internal Flag is 20.
Code is
0 for Disable
1 for Enable
To enable Tape calls coming without CLI Flag in a Station Advanced Feature Template, dial:
• 5602-1-Template Number-Feature Number-Code
Where,
Template Number is from 01 to 50.
Feature Number for Tape calls without CLI is 17.
Code is
0 for Disable
1 for Enable
• Exit SE mode.
For SE commands for applying the programmed template to DKP and SLT extensions, refer the topic
“Customizing Station Advanced Feature Template using a Telephone”.
How to use
This feature works automatically on extensions which have the related Call Taping parameters programmed in their
Station Advanced Feature Template.
Call Taping conversations are recorded in a single, common mailbox. These can be accessed directly by the
Mailbox Owner (user of the extension to which this common mailbox is assigned). Other extension users can also
access this common mailbox by calling the Voice Mail System.
OR
If the common mailbox is password protected, make sure that you provide the password to all extension
users who are to be provided access to this mailbox.
The above instructions contain the default access codes. Check with your System Engineer, if these have
been changed and use the current access codes.
142. This is the default Voice mail Feature Access Code. Verify with you System Engineer if this has been changed and use the new
code.
143. Only if the mailbox is password protected, you will be prompted to enter the password.
144. Only if the mailbox is password protected, you will be prompted to enter the mailbox password.
What's this?
Call Toggle allows you to have two simultaneous telephone conversations, talking to two persons alternately.
How it works
• A, B, and C are extensions.
• A is in speech with B.
• A wants to talk to C.
• A is in speech with B.
• A wants to talk to D.
• A is in speech with D.
• A wants to talk to E.
• A gets Ring Back Tone and D is put on hold. D gets music on hold.
• The party put on hold during Call Toggle cannot hear the conversation between the other two parties.
• You can also toggle between an incoming internal/external call (indicated by call waiting tone) and an
internal/external call you are currently in speech with.
• You can also answer an incoming 'Interrupt Request' call and toggle between the interrupting extension
and the extension you were in speech.
• You can convert a Call Toggle into a three-party conference by dialing Flash-0.
• You can transfer the call you are currently in speech with to another extension.
• You can park the call you are currently in speech with.
How to configure
Call Toggle is a Class of Service (CoS)-dependant feature.
In the default Station Basic Feature Template 01 assigned to all extensions of ETERNITY, Call Toggle is included in
the 'Basic Features' assigned to all CoS groups, including the default CoS group 01. So, all extensions of
ETERNITY can use this feature.
As Call Toggle is a part of the set of 'Basic Features', you cannot disable this feature selectively in the COS of
extensions, without disabling the entire set of features.
No specific programming is required for this feature, except for programming a DSS key for Call Toggle, if required.
Refer the topic “DSS Keys Programming” for instructions.
OR
• Dial Hold-1.
• Speech with extension 1.
• Press DSS key assigned to Call Toggle again
OR
• Dial Hold-1.
• Speech with extension 2.
OR
• Dial Hold-1.
• Speech with extension.
• Press DSS Key assigned to Call Toggle
OR
• Dial Hold-1.
• Speech with external party.
• Dial Hold-1.
• Speech with external party 1 on trunk 1.
• Press DSS Key assigned to Call Toggle again.
OR
What is this?
Call Transfer enables you to relocate an existing call from an extension or trunk to another extension or to an
external number. Calls can be transferred after notifying the other extension/external number about the impending
transfer or can be transferred directly without notification.
• Call Transfer - Screened: The Operator puts the caller on hold, dials the desired party's extension, and
informs the desired party of the impending transfer. If the desired party chooses to accept the call, the call
is transferred over to them.
• Call Transfer - While Ringing: The Operator puts the caller on hold, dials the desired party's number and
transfers the call when the desired party's extension starts ringing.
This feature is used when there are several other calls to be attended and the Operator cannot wait for the
desired party to answer.
• Call Transfer - On Busy: The Operator puts the caller on hold, dials the desired party's number and
transfers the call even when the desired party is busy in speech with another person. The busy extension
gets intrusion tone and can choose to answer the intruding (transferred) call.
• Call Transfer - Trunk-to-Trunk: An external call is transferred on to another trunk line. The Operator puts
the external caller on hold, dials the desired party's external number, and transfers the call after or without
notifying the desired party of the impending transfer.
Trunk-to-Trunk call transfer may be used to transfer incoming calls for out-of-office extension users to their
cell phones, or to connect personnel at remote or distant locations. For instance: an out-of-office executive
who does not have long distance dialing permission can call the office and request the operator to connect
him to the desired party on a trunk line.
• Blind Transfer to VMS: The Operator puts the caller on hold, dials the feature access code for Blind
Transfer to VMS, dials the desired party's number, and transfers the call. The call is transferred to the
mailbox assigned to the desired party. The caller may leave a message in the mailbox.
Call Transfer is not exclusively an Operator feature, though it is used mostly by Operators. Calls can be
transferred by any extension to another extension or external number, if "Basic Features" are allowed in
Class of Service of the transferring extension.
How it works
A and B are extension users.
C is an external caller.
D is an external number.
1. Screened Transfer:
• If B does not accept the call, Operator may dial Flash to retrieve the call and speak to C.
• The Operator can also abort call transfer while B's phone is ringing by dialing Flash. The Operator gets
connected to C.
3. Transfer On Busy:
• If B does not answer the intrusion beeps at the end of the Transfer on Busy Timer, the call is returned
to the Operator. C gets ring back tone.
• If the Operator too is busy at the time of call return, C gets busy tone.
OR
Transfer the call as soon as D's phone starts ringing. (transfer while ringing)
• C and D are now in speech for the duration of the Trunk-to-Trunk Inactivity Timer145.
• A warning tone is given at the end of the Trunk-to-Trunk Inactivity timer (programmable; default: 90
seconds). On expiry of this timer, the call is disconnected.
• To extend the call, either C or D must dial any digit in tone (DTMF), except '##'.
In the case of Trunk-to-Trunk transfer, both parties in speech on trunk lines must be informed that their call
would be disconnected at the end of the Trunk-to-Trunk Inactivity Timer and that they must dial any digit,
except ‘##” to extend the call.
• If A does not have a mailbox assigned, the Operator will get an error tone while transferring the call.
• The Operator may retrieve C's call by pressing Hold/Flash/Call Appearance key.
Feature Interactions:
• CLIP and Caller ID Presentation while Transfer: ETERNITY provides the flexibility to display either the
extension number that is transferring the call or the held party's number, i.e. the number of the party that is
about to be transferred. Refer “Calling Line Identification and Presentation (CLIP)”.
• Privacy: Call Transfer-On Busy will not work if the busy extension has Call Privacy from intrusion Tone in
its Class of Service.
• DND: Call Transfer will not work if the destination extension has set DND.
145. The process of Trunk-to-Trunk transfer takes place outside of the PBX. So, the PBX will not know which of the two trunks have
gone ON-Hook. Hence the Trunk-to-Trunk Inactivity Timer. The call is automatically disconnected when this timer expires.
To be able to use Blind Transfer to VMS, the extensions must be assigned a mailbox in the VMS of the system.
Refer the System Manuals for the VMS Card and CadencePro to know more about this and for programming
instructions.
You cannot disable 'Call Transfer' selectively without disabling the entire set of 'Basic Features'.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” to know more.
• Transfer on Busy Timer: This timer is related to Call-Transfer on Busy. It is the time for which the system
waits for the busy extension to respond to the intrusion tone. By default the timer is set to 30 seconds. At
the end of the timer the call is returned to the transferring extension.
• Trunk to Trunk Inactivity Timer: This is the time duration after which the system disconnects the call
transferred from one trunk line to another. By default it is set to 90 seconds. At the end of the timer the call
is disconnected, if either party does not dial digits to extend the call. This Timer is relevant for TWT to TWT
and TWT to E&M calls only.
• Scroll to reach the Call Transfer related timers and change the values as required.
• Exit SE mode.
How to use
OR
OR
• Go ON-Hook.
Extension to Trunk:
OR
• Press Hold.
• Dial-Trunk Access Code146-External Number and go On-Hook.
OR
• Press Hold and dial desired party's extension number and Go-ON Hook.
146.Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.
OR
OR
• Dial 1078.
• Dial desired party's extension number.
• Go Idle or you get dial tone after 3 seconds.
Extension to Trunk:
147.Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.
What's this?
The ETERNITY provides the facility of detecting the caller's number and presenting it on the display of the called
extension phone. This feature is called Calling Line Identification and Presentation (CLIP).
The calling number can be presented on ISDN Terminals, EON and also on SLTs that support CLI protocols.
The signaling protocols for CLI supported by ETERNITY are: DTMF, FSK V.23, and FSK-Bellcore.
These protocols are supported on trunks as well as extensions. Any type of trunk line and supporting DTMF or FSK
signaling can be interfaced with the ETERNITY.
Similarly, any type of telephone instrument supporting DTMF or FSK signaling protocol can be connected to the
SLT port.
How it works
When CLIP is enabled on a trunk,
• It sends this information to the landing extension/Operator along with the ringing signal.
• In case of, Internal calls the calling extension's name and number both are presented to the called
extension.
• In the case of External calls, only the number will be displayed on the landing/Operator extension.
• When the landing extension/Operator transfers the incoming call to an extension, putting the external
caller on hold, the system sends this information to the extension to which the call is transferred.
• During the transfer, the number of the landing extension/Operator will be displayed on the transfer
destination extension.
• On successful call transfer, the caller's number will be displayed on the transfer destination extension.
In the case of Call Transfer, the system also provides the option of displaying to the destination extension either the
number of the party that is put on hold to be transferred, i.e. the Held Party OR the number of the Transferring
Party, while the call transfer is taking place. This feature is called Caller ID Presentation while Transfer.
It is also possible to remove and replace the '+' character received as CLI on telephones that do not support CLIP
starting with this character.
For example, the GSM network sends the calling party number with '+' as the prefix. If the telephone connected as
extension does not support this, it will not present the CLI of the caller. To overcome this, ETERNITY provides you
the option of replacing '+' with an appropriate number string which these telephones can display.
• CLIR: CLIP and Caller ID Presentation while Transfer will work only if CLIR is not enabled on the
extension that has transferred the call. Refer the topic “Calling Line Identity Restriction (CLIR)”.
• Q-Sig: When two PBXs - PBX A and PBX B are networked using Q-Sig, and an extension of one PBX, for
instance, PBX A transfers a call to an extension of PBX B by putting the caller on hold, the CLI presented
on the extension of PBX B will be according to the type of Caller ID Presentation while Transfer set on the
transferring extension of PBX A.
How to configure
The functioning of this feature is controlled by two parameters: 'CLIP Type' and 'Caller ID Presentation while
Transfer'.
If you want to replace '+' characters received as CLI on telephones that do not support CLI prefixed with this
character, you must program the relevant flag and the desired number string in the 'System Parameters'.
CLIP Type
If SLTs supporting CLI are connected to the ETERNITY, the System Engineer must select a signaling protocol for
CLI in the SLT Hardware Template applied on the SLT extensions. By default SLT Hardware Template 01 is
assigned to all SLT extensions. The default CLIP Type in Template 01 is 'DTMF'.
There is no need to select a CLIP Type in the default Hardware Template 01, if all the SLTs support DTMF protocol.
If all SLTs support a different CLIP Type say FSK-Bellcore, you may simply select this CLIP Type in the default
Hardware Template applied on all SLT extensions.
However, if certain SLTs support a particular CLIP type and some support a different CLIP type, then create
separate SLT Hardware Templates with different CLIP types and apply them to the appropriate SLTs.
For example, you may select SLT Hardware Template 02 with FSK V.23 and SLT Hardware Template 03 with FSK
Bellcore, and Template 04 with DTMF as the CLIP Type and apply each template to the SLTs as per the CLIP Type
they support.
• Exit SE mode.
This feature is to be programmed in the Station Advanced Feature Template applied on the extension.
In the Station Advanced Feature Template 01 assigned to all extensions by default, Caller ID of the Held Party is
selected.
But if you want to show the Caller ID of the Transferring party, select this option in the default Station Advanced
Feature Template 01.
If all certain extensions are to be provided Caller ID of the Held Party and others the Caller ID of the Transferring
Party,
• Exit SE mode
Refer the topic “Station Advanced Feature Template” for instructions assigning Templates to stations.
• Enable the 'Replace '+' CLI?' flag by selecting the check box.
• Enter the desired number string in the field 'Replace '+' from CLI with the number string'.
• Exit SE mode.
What's this?
The ETERNITY allows extension users to suppress their identity, i.e. extension number and name, when they call
other extensions. This feature is called Calling Line Identity Restriction (CLIR).
Extensions that have 'CLIR Override' facility can view the CLI of those that have suppressed it with CLIR.
This is a feature of the PBX and not the PSTN. It is applicable for internal calls only.
This feature will work only on the proprietary digital key phone EON, the Matrix Extended IP Phone, and SLTs that
support Caller Line Identification (CLI).
How it works
• Extension A has CLIR enabled.
• Extension B does not have CLIR Override enabled.
• Extension C has CLIR Override enabled.
• When A calls B, B cannot view the extension name and number of A.
• When A calls C, C can view A's extension name and number.
Now,
• Extension D calls extension E.
• A picks up the call.
• D will be able to view A's name and extension only if it has CLIR Override enabled and is a digital key
phone, EON.
Feature Interactions:
• CLIP and Caller ID Presentation while Transfer: Both these features will not work if CLIR is enabled.
How to configure
'CLIR' and 'CLIR Override' are Class-of-Service-dependant features. Extensions that are to be allowed these
features, must have them enabled in their Class of Service (CoS) group.
Decide which extensions should be allowed CLIR and which should be allowed CLIR Override.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. Template 01 is assigned CoS group 01 in which both CLIR and CLIR Override are disabled. Thus,
none of the extensions of the ETERNITY can suppress their CLI or force any other extension to display its CLI.
If you want to enable both features on all extensions, simply enable CLIR and CLIR Override in the default CoS
group 01.
If you want to allow CLIR to all extensions, but not allow CLIR Override to any extension, simply enable CLIR in the
default CoS group 01.
If you want to allow CLIR and/or CLIR Override to selected extensions, only, then follow these steps:
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the extensions to which CLIR/CLIR Override is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
How to use
OR
To disable CLIR:
OR
• Dial 1030.
• You get confirmatory tone and message on the phone's display.
• Go idle or you get dial tone after 3 seconds.
• Lift handset.
• Dial 103-1.
• You get confirmation tone.
• Replace handset.
To disable CLIR:
• Lift handset.
• Dial 103-0.
• You get confirmation tone.
• Replace handset.
148. System Engineer is recommended to assign a DSS Key with LED to this feature. When the assigned DSS key is pressed, it will
glow red indicating that CLIR is enabled.
149. If a DSS key with LED has been assigned, when you press the key again, the LED will be turned off indicating CLIR is now
canceled.
What's this?
For each feature of the ETERNITY that an extension user has set/enabled on his/her extension, the system
provides access code for cancellation of the feature.
However, there is also a single master command for extension users with which all features that are set on an
extension can be canceled.
When the extension user dials 'Cancel All Station Features' command, the following features, if set, are cancelled
from the extensions:
• Alarms
• Auto Answer
• Auto Call Back
• Auto Redial
• Background Music
• Call Follow-Me
• Call Forward
• Do Not Disturb
• Hot Line
• Hot Outward Dialing
• Trunk Reservation
• Walk-In Class of Service
Auto Redial, Background Music, Call Forward and Call Follow-Me, Do Not Disturb, Hotline and Trunk
Reservation are Class of Service dependent features. These features can be set/canceled from an
extension only if they are included in its “Class of Service (COS)”.
How to use
OR
• Dial 1051.
• You get confirmation tone and confirmatory message on your phone display.
• Go idle or wait for dial tone.
What's this?
Class of Service (CoS) defines the permission an extension will have on a PBX. It defines the set features of the
PBX that the extension is to be allowed access to.
Feature requirements vary among users and with time. Certain groups of extension users may have a need for
voice mail, while another group may need the ability to forward calls to a cell phone, and still others may have no
need to make calls outside the office.
Similarly, certain features that are required during working hours may not be required during break or non-working
hours.
ETERNITY offers the flexibility to allow or deny extension users access to features of the PBX, on the basis of their
requirement and according to time of the day. For users, access to various features from their extensions is their
CoS.
How it works
The list of all features allowed to an extension is referred to as 'CoS group'. There are 20 CoS groups numbered
from 01 to 20.
In each CoS group there are 53 features, which are identified by 2-digit numbers, from 01 to 53. These are referred
to as the 'CoS Feature Numbers'.
Each station port of the PBX has an associated CoS group that indicates which features of the PBX the port is
allowed to access.
The CoS group of a station port is defined in the “Station Basic Feature Template” applied to that station port. It is
defined for each "Time Zone", namely, working hours, break hours, and non-working hours, in the Template.
A feature can be allowed or denied to an extension by enabling or disabling it in the CoS group of the Station Basic
Feature Template applied to that extension.
The same CoS group uniformly to all stations ports for all Time Zones. Doing so, all extensions can access the
same set of features in all time zones. For example: CoS group 03 is assigned to all extensions for Working, Break
and Non-Working hours.
A different CoS group for each Time Zone can be assigned to all station ports. Doing so, all extensions can access
only those features allowed for the particular Time Zone.
For example: All extensions are assigned CoS group 03 for Working, CoS group 04 for Break hours and CoS group
05 for Non-Working Hours.
Different CoS groups can be assigned to different station ports, for all or for different Time Zones. Doing so, each
extension can access a different set of features in each Time Zone.
Basic Features
A set of features including Internal Call, Call Hold, Call Toggle, Call Transfer, Department Call, Operator Access,
Redial, and Call Mute defined as Basic Features and allowed in all CoS groups.
It is not possible to enable or disable selectively any of the features included in "Basic Features".
How to configure
The table below presents the CoS groups from 01 to 20 with the list of 01 to 53 features supported on the stations.
• CoS group number 01 is assigned for all Time Zones in the default Station Basic Feature Template 01
assigned to all stations of the ETERNITY.
• CoS group number 19 and 20 are assigned when the Hospitality Application of ETERNITY is used. See
ETERNITY Hospitality System Manual.
• Read the list of features supported on the extensions (see above Table 'Default CoS Groups for
Features').
• Against each extension name on the list, write the features needed for each Time Zone. You will notice
that the features needed by many extensions are identical.
• List the common features to be allowed to and features to be denied to all extensions. Assign a CoS Group
Number to this list.
• Are there any other features, in addition to those on the common list, which you want to allow to selected
extensions?
• If yes, extend the common list you prepared by adding the features to be allowed to selected extensions.
Assign a CoS Group Number to this extended list.
• When you are finished preparing the CoS groups you need, program the CoS groups using Jeeves or by
issuing SE commands from a Telephone (DKP or SLT). See below for instructions.
• Now, the CoS groups to be assigned to extensions must be programmed in the Station Basic Feature
Template applied to the extensions. This can be done using Jeeves or by issuing SE commands from a
Telephone (DKP or SLT). See below for instructions.
• The default CoS groups from 01 to 20 appear. The check boxes selected under each CoS group column
indicate that the feature is enabled in that CoS group. The default CoS groups meet the requirements of
most extension users. Check the default CoS groups whether the features you want to allow are enabled
and features you want to deny are disabled.
• Exit SE mode.
• The default CoS group assigned to each time zone, i.e. working hour (WH), non-working hour (NH) and
break hour (BH), appears under Class of Service in each Template.
• To assign a CoS group to a Station Basic Feature Template, enter the CoS group number for each time
zone under Class of Service.
By default, Station Basic Feature Template 01 is applied on all stations and CoS group 01 is the assigned
by default to this template in all time zones.
If all stations to be allowed the same set of features during working hours, break hours, non-working hours,
enter the same CoS group number in all time zones in the Template Number applied to all stations. For
• If a set of features is to be allowed to select stations only, assign the CoS group with these features
enabled to a separate Station Basic Feature Template. Apply this template to the select stations which are
to be allowed this CoS.
For example: To assign all features to extensions, create a CoS group with all features enabled, CoS
group 07. Select a different Station Basic Feature Template, for example 05. Enter CoS Group 07 in all
• Remember to click 'Submit' to save the changes you make on every page.
For Example: To program CoS group 04 in Station Basic Feature Template Number 06 for all Time
Zones, dial:
• 5502-1-06-03-04 to program CoS group 04 for Working Hours.
• 5502-1-06-04-04 to program CoS group 04 for Break Hours.
• 5502-1-06-05-04 to program CoS group 04 for Non-Working Hours.
Similarly, to program CoS Group 03 in working hours, 04 in Break Hours and 05 in Non-Working hours,
dial:
• 5502-1-06-03-03 to program CoS group 03 for Working Hours.
• Exit SE mode.
After you have programmed the CoS group in the Station Basic Feature Template, you must assign this
template to the stations. Refer the topic “Station Basic Feature Template” for instructions on applying
templates on SLT, DKP, ISDN Terminal and SIP extensions.
Finally, test the CoS programmed for each extension by invoking the features from each extension.
What is this?
ETERNITY offers the facility to detect the calling party's number and name. This is known as Calling Line
Identification.
On the basis of CLI, it is possible to land calls from a particular telephone number on a particular extension or group
of extensions. This is known as CLI Based Routing.
How it works
A, B, C are extensions. D and E are external callers.
Calls made by D are to be landed on A
Calls made by E are to be landed on B and C.
The CLI of D and E and their corresponding landing destinations should be entered in the CLI Based Routing
Table.
If D's number does not exist in the CLI Based Routing Table, the call will be routed according to the incoming
call management logic.
How to configure
For this feature to work, you must enter the numbers of the calling parties and the numbers of the corresponding
destination extensions in the CLI Based Routing Table. You can store up to 400 numbers in the CLI Routing Table.
: :
• At each Location Index, enter the information for the following parameters:
• Number: enter the number of the calling party, not exceeding 16 digits. You can also enter '+' in the
number string.
• Name: enter the name of the calling party. You can enter a maximum of 8 characters in this field.
• Port Type: select the landing destination extension. It may be an SLT, a DKP, an ISDN Terminal, SIP
extension, or a Routing Group.
• Port Number: enter the software port to which the landing destination SLT/DKP/ISDN Terminal/SIP
Extension is connected. If you have selected a Routing Group as the landing destination, enter the
number of the Routing Group (01 to 96) in this field.
To add the calling party telephone number in the CLI Based Routing table, dial:
• 4101-Index-Telephone Number-#*
Where,
Index is from 001 to 400.
Telephone Number is the calling party's telephone number (Max. 16 digits).
Terminate the command with #*, if the number is less than 16 digits.
To enter '+' in the number string, dial #8
To clear a telephone number from the CLI Based Routing table, dial:
• 4101-Index-#*
To enter the name of the calling party corresponding to the calling party's telephone number, dial:
• 4102-Index-Name-#*
Where, Index is from 001 to 400.
Name can be a maximum of 8 characters.
Terminate the name with #*, if less than 8 characters.
For Example: to enter the name Midas Biz for the number 2640459 at Index Location 001, dial 4102-
001-MidasBiz
To clear the name stored at a location index in the CLI table, dial:
• 4102-Index-#*
For Example: to assign an SLT connected at Software Port number 004 as the landing destination for
calls from Midas Biz 2640459, dial 4103-001-01-004
By using these commands, the telephone number, name and the Port type-Port number will be cleared
from the location index.
• Exit SE mode.
What's this?
When data is transmitted from the ETERNITY to external lines or when ETERNITY receives data from the external
lines, it is necessary that the transmitter and receiver be properly synchronized. If not clock slips can occur. A clock
slip can generate a loss or addition of data to the data stream.
How it works
• This can be done in three ways viz. using the data clock or using the external clock (clock is sent by the
network on a dedicated cable pair) or using the internal clock. ETERNITY does not support external clock.
When the ETERNITY is connected to the PSTN, then it is recommended to extract the clock from the
incoming data whereas if the ETERNITY is used to form a private network, you are recommended to use
the internal clock. For example, if a private network is formed by connecting three ETERNITY systems,
then one system should be programmed as master clock whereas other two should be programmed in the
slave mode.
• If two or more T1E1 Ports are connected to the PSTN (or a Private Network) then in such case, clock will
be extracted from the first T1E1 Software port whereas the transmit data on all other ports whether
connected to PSTN or private network will be clocked as per received.
How to configure
05 T1E1 1-8
04 BRI 01-32
00 Null 000
1 T1E1-1
2 T1E1-2
3 T1E1-3
4 T1E1-4
The system checks this table for a master lock. If none of the ports is synchronized out of this table, the system
gives priority to the internal clock. If any one port is synchronized, the system selects that port as a system clock
master. Here index is given priority, that is, if the second port of this table is selected as clock master and suddenly
first port is synchronized, then the system changes its master from 2nd port to first port. Now if first port has lost its
synchronization then in this case again, the second port is selected as master clock of the system.
1 8 KHz Derived
2 8 KHz
3 2.048 MHz
4 1.54 MHz
By Default, 'System Clock Synchronization' is 2.048 MHz for India and other countries except USA.
For USA, default 'System Clock Synchronization' is 1.54 MHz.
• This command is applicable only for 'ETERNITY GE', when Software version/revision 'V8R6' is used.
• Selecting option 1('8 KHz Derived'): If you are using, the Software version/revision 'V8R6' onwards,
with CPLD version/revision 'V1R2 or earlier', the System Clock Synchronization will be done only at '8
KHz Derived' option, irrespective of the selected 'System Clock Synchronization' option. If you are
using the BRI/T1E1 card with CPLD version/revision 'V1R3 and onwards', the clock synchronization
option will work as you have programmed the option, using this command. To know the CPLD version/
revision, open the cover of T1E1 card of your system and check the label on the CPLD device or
contact your dealer for more information.
• Selecting option 2('8 KHz'): By default, Master Clock Synchronization port number is given as T1E1-1,
T1E1-2, T1E1-3 and T1E1-4. The options for Master clock synchronization allow selecting T1E1 or BRI
port or a combination thereof as required. If any BRI port is selected in Master Clock Synchronization
option (for any option from 1 to 4), SE should select the System Clock Synchronization option 2 = 8
KHz.
• Selecting option 3('2.048 MHz'): Select 'System Clock Synchronization' option 3 = 2.048 MHz, only for
E1 T1E1 line.
1 Fast
2 Slow
What’s this?
To have private networks, few PBXs can be connected to each other using E&M, T1/E1, QSIG, etc. The
requirement demands that the PBXs connected to each other forming the network behave as a single group.
The users need not dial a separate code to access a station user of other PBX. The entire network should
behave as a single unit. Users will not know whether they are dialing an extension number of their own PBX or
of the other PBX. This is called Closed User Group.
PBX-A PBX-B
T1 E&M1 T1
T2 T2
PSTN PSTN
E&M2 E&M3
Tn Tn
S1 S2 Sn S1 S2 Sn
PBX-C
S1 S2 Sn
In the above figure, 3 PBX systems are connected through E&M connectivity.
• S1 to Sn are stations.
• E&M1 to E&M3 are E&M lines between the three PBX systems.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How it works
For Closed User Group, it is mandatory to have unique station number in all the systems i.e. one cannot have
station number 2001 in PBX-A as well as in PBX-B or PBX-C.
Few new words have been used to explain this application, each of these words have been explained below:
Route Index Route Code OGTBG Strip Digit Count Self Router Flag Max. Dialed Digits
001
002
003
250
• Route Code: Route code could be of maximum sixteen digits. Digits 0 to 9 are allowed. However, ‘*’ and
‘#’ are not allowed. Generally route code will be a truncated number of the station numbers. For example in
the figure given above, route code for PBX-B can be defined as ‘3’ and that for PBX-C can be defined as
‘4’.
If PBX-B were having station numbers from 3100 to 3199 and PBX-C were having station numbers from
3200 to 3299 then route code for PBX-B can be defined as ‘31’ and that for PBX-C can be defined as ‘32’.
If PBX-B were having station numbers from 301 to 399 and 401 to 499 then two route codes can be
defined for PBX-B viz. ‘3’ and ‘4’. Likewise for PBX-C.
• OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the station
number is dialed on it. The same logic of Rotation On/Off for trunk selection from the OGTBG is used. If
rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next trunk in the
group is selected. This helps to select an alternate route. Whereas if Rotation is ON then the trunks in the
OGTBG are selected in round robin fashion.
• Strip Digit Count: It has no significance for Closed User Group application. But it has to be programmed
as 0.
• Self-Route Flag: It has no significance for Closed User Group application. But it has to be programmed as
0.
• Maximum dialed digits: When digits are dialed on the trunk, the system waits for inter digit timer after the
last digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further delay,
count for the number of digits to be programmed in this field. If the number of digits received are equal to
the parameters programmed then the number is dialed out immediately without waiting for the inter digit
timer. If the number of digits dialed by the user are not equal to the digits programmed, the number is
dialed after inter digit timer.
The ETERNITY has only one routing table. The same table is used for Closed User Group and Close User
Group-With Exchange ID applications. Hence the table has to be programmed keeping the application in
mind.
Step 1
Use following command to configure route code:
4502-1-Route Index-Route Code-#*
4502-2-Route Index-Route Index-Route Code-#*
4502-*-Route Code-#*
Where,
Route Index is from 001 to 250.
Route Code is a sixteen digits string of numbers.
Step 2
Use following command to assign OG Trunk Bundle Group to the route code:
4503-1-Route Index-OG Trunk Bundle Group
4503-2-Route Index-Route Index-OGTBG
4503-*-OG Trunk Bundle Group
Where,
Route Index is from 001 to 250.
OG Trunk Bundle Group is from 01 to 32.
By default, OG Trunk Bundle Group is 01.
Step 3
Use following command to configure strip digit count for a route:
4504-1-Route Index-Strip Digit Count
4504-2-Route Index-Route Index-Strip Digit Count
4504-*-Strip Digit Count
Where,
Router Index is from 001 to 250.
Strip Digit Count is from 0 to 9.
By default, Strip Digit Count is 0.
Step 4
Use following command to configure self-route flag for a route:
4505-1-Route Index-Code
4505-2-Route Index-Route Index-Code
4505-*-Code
Where,
Code Meaning
Step 5
Use following command to configure maximum dialed digits to select router for a route code:
4506-1-Route Index-Maximum Dialed Digits
4506-2-Route Index-Route Index-Maximum Dialed Digits
4506-*-Maximum Dialed Digits
Where,
Route Index is from 001 to 250.
Maximum dialed digits is from 00 to 99.
Step 6
Use following command to clear an entry in a routing table:
4501-1-Route Index
4501-2-Route Index-Route Index
4501-*
Where,
Route Index is from 001 to 250.
ETERNITY offers few features associated with closed user group. Each of these are discussed below:
Alternate Route:
• This feature provides flexibility of accessing a station of other exchange through alternate routes if the
normally used route (the shortest route) is not free. To achieve this, the trunks that offer shortest routes
should be programmed first in the OGTBG followed by the trunks that provide alternate routes. Also the
rotation within the OGTBG should be OFF. In figure the requirement is that if E&M1 is busy then E&M2
should be used to call 3001 from PBX-A. In this case a OGTBG is to be so formed that it has E&M1 as
first trunk and E&M2 as second trunk and should be assigned to route code ‘3’. However, the rotation
within the OGTBG should be disabled. Similarly, if the call is to be made to 4001 then E&M2 should be
used. Hence another OGTBG should be programmed with E&M2 as first trunk and E&M1 as second
trunk and it should be assigned to route code ‘4’. Also the round robin option for the OGTBG should be
selected.
Transit Barring:
• This feature helps to bar the Transit calls through the exchange. Consider figure 1. It is required that a
station user in PBX-A can access station 3001 using alternate route through PBX-C but a station user
in PBX-B cannot access stations 2001 to 2010 using alternate route through PBX-C. This can be
accomplished using Transit Barring. To achieve this, a denied list containing 10 numbers viz. 2001 to
2010 should be assigned as Toll Control for the SLT port programmed for the E&M2. Doing so when a
station user from PBX-B dials 2001, if E&M1 is busy, the system would try dialing through E&M2 but
since E&M2 does not have requisite toll control, it will give error tone to station user. Transit Barring
adds value to the Alternate Route by allowing a selective access to the station with alternate route.
• Please refer figure 1. When a station in PBX-A dials a station number 3001, the system searches for
this number in PBX-A. Since there is no station with flexible number 3001 in PBX-A, the system
checks the E&M routing table. The system follows the routing table, identifies that the dialed number is
in PBX-B, selects a free E&M path and reaches the dialed port.
• As shown in figure 1, the shortest path to reach station 3001 from PBX-A is through E&M1. But if E&M1
is busy then the network can be programmed to reach station 3001 through E&M2 and E&M3. For this
both PBX-A and PBX-B have to be programmed to accomplish this.
Relevant Topics:
1. “E&M Connectivity” 1203
2. “Forced Call Disconnection” 1234
3. “Closed User Group-With Exchange ID” 1033
4. “OG Trunk Bundle Group” 1352
What’s this?
• To have private networks, few PBXs can be connected to each other using E&M, T1/E1, QSIG, etc. The
requirement demands that the PBXs connected to each other forming the network behave as a single
group. The users need not dial a separate code to access a station user of other PBX. The entire network
should behave as a single unit. The extension users will not know whether they are dialing an extension
number of their own PBX or other PBX. This is called Closed User Group. However, it is possible that the
PBXs connected to form a network may have same station numbers. Also, all the exchanges within the
network may have their own identity (called Exchange ID). In such cases, the routing scheme (the routing
table) has to be programmed keeping the Exchange ID (EID) in mind. This is known as Closed User
Group-With Exchange ID.
• This facility is generally used in PLCC Applications wherein new power stations (and hence PBXs) are
added in the network. It is not feasible to have unique station numbers throughout the network. In such
cases, an Exchange ID is assigned to the newly added PBX and a routing table is programmed in the
exchange. Also, the routing tables of other exchanges are modified to include the newly added exchange
in the network.
In the below figure, 3 PBX systems are connected through E&M connectivity.
• S1 to Sn are stations.
PBX-A PBX-B
T1 21 E&M1
22 T1
T2 T2
PSTN PSTN
E&M2 E&M3
Tn Tn
S1 S2 Sn S1 S2 Sn
PBX-C 23
S1 S2 Sn
• Routing Table: This table has five parameters viz. Route Index, Route Code, OG Trunk Bundle Group,
Strip Digit Count and Self Route flag. The Closed User Group-With Exchange ID programming works
according to this table.
Route Index Route Code OGTBG Strip Digit Count Self Router Flag Max. Dialed Digits
001
250
• Route Code: Route code could be of maximum six digits (XXXXXX). Digits 0 to 9 are allowed. Generally,
route code will be a unique number. The route code should not clash with any of the station numbers of
same PBX. For example in the figure given above, route code for PBX-A can be defined as ‘21’, route code
for PBX-B can be defined as ‘22’ and that for PBX-C can be defined as ‘23’. This means that no station in
PBX-A can start with ‘22’ or ‘23’. Similarly, no station in PBX-B can start with ‘21’ or ‘23’ and no station in
PBX-C start with ‘21’ and ‘22’.
• OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the station
number is dialed on it. The same logic of rotation On/Off for trunk selection from the OGTBG is used. If
rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next trunk in the
group is selected. This helps to select an alternate route. Whereas if rotation is ON then the trunks in the
OGTBG are selected in round robin fashion.
• Strip Digit Count: This count signifies the number of digits to be stripped off while dialing/decoding a
number. To elaborate: Consider figure 1. The requirement is that if station 2001 of PBX-B dials 212002
and if E&M 1 is busy then the call should reach station 2002 of PBX-A through alternate route. In this case
the strip digit count of PBX-A should be programmed as 2 and that of PBX-B and PBX-C should be
programmed as 0. Doing so, when station 2001 of PBX-B dials 212002 and if E&M1 is busy then the call is
routed through PBX-C. In this case, PBX-B dials 212002 on E&M3, PBX-C receive this code and dials out
the same code i.e. 212002 on E&M2 without striping of any digit. On receiving 212002, PBX-A strips of two
digits as per the programming and routes the call to station 2002.
• Self-Route Flag: This flag signifies that the digits being dialed are for the same PBX and are not to be
dialed on the E&M trunk.
• Maximum dialed digits: When digits are dialed on the trunk, the system waits for inter digit timer after the
last digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further delay,
count for the number of digits to be programmed in this field. If the number of digits received are equal to
the parameters programmed then the number is dialed out immediately without waiting for the inter digit
timer. If the number of digits dialed by the user are not equal to the digits programmed, the number is
dialed after inter digit timer.
How to configure
Please refer topic “Closed User Group (CUG)” for more details.
ETERNITY offers few features associated with Closed User Group-With Exchange ID. Each of these are
discussed below:
Alternate Route
Please refer “Closed User Group (CUG)” for more details.
Transit Barring
Please refer “Closed User Group (CUG)” for more details.
Strip Digit Count is of significance in a network in which few exchanges possess Exchange ID whereas
others do not. Refer figure 2. PBX-A and PBX-B are made to work as Closed User Group since they have
unique stations. But since PBX-C possess stations whose flexible number clashes with stations in PBX-A,
it cannot be made a part of Closed User Group. In such case Closed User Group-with Exchange ID can be
used with the combination of PBX-A + PBX-B and PBX-C forming the network.
In this case, if 2001 of PBX-A wants to access 3001 through E&M1, he must dial 3001. But if E&M1 is busy
then system will allot him E&M2 and since there would not be any programming done, it will give error
tone to the caller. To avoid this condition, the station user of PBX-A should be asked to call 3001 by dialing
223001. For this, the strip digit count for the route index with route code ‘22’ in the routing table of PBX-B
should be programmed as ‘2’ and the strip digit count for the route index with route code ‘22’ in the routing
table of PBX-C should be programmed as ‘0’. Doing so, if the call to 3001 is made through E&M1, then
PBX-B would strip of the first two digits on receiving 223001 and make the caller reach 3001. If the call to
3001 is routed through E&M2, the PBX-C will not strip of any digit and would dial out 223001 on E&M3. On
receiving 223001, PBX-B as per the programming would strip of 22 and make the call land on 3001.
•‘Prefix String’ is a string of characters which is prefixed to the string, dialed by the user and then CUG
Routing-Table is applied.
Example:
• If this feature is not programmed for Exchange of type: ET1 or BPL, and ‘Prefix String’ is blank, then, only
‘2223’ is dialed by the Exchange when ‘02223’ is dialed by the user, to call station ‘23’.
• Now when ‘2223’ reaches Exchange B, since there is no entry in the CUG table, the feature-extension
flexible number table is checked. Now since a match starting with ’22..’ is found, the call is routed to station
‘22’ instead of ‘23’, because station number ‘22’ is present on the Extension.
• To avoid this, an entry is made in the routing table containing ‘22’ as the route code with strip digit count
‘2’. But then since the system checks CUG Routing-table first, the first two digits out of ‘2223’ always get
striped off and the call is not routed to station ‘22’. i.e. the other stations of Exchange B will never be able
to call station ‘22’.
• To solve this problem ‘Prefix String’ feature is programmed in the E&M Feature Template with ‘0’ as prefix
string, so that ‘string with prefix’ (022) is matched with the entries of CUG Routing Table (as shown in
figure). When user dials ‘02223’, first 3-digits are stripped off and the required Extension ‘23’ can be
called.
Relevant Topics:
1. “E&M Connectivity” 1203
2. “Forced Call Disconnection” 1234
3. “Closed User Group (CUG)” 1028
4. “OG Trunk Bundle Group” 1352
5. “E&M Feature Template” 565
What's this?
The CO Call Waiting feature gives indication to the user of a busy extension about the waiting call on a Trunk.
This is a “Class of Service (COS)” dependent feature. Only those extensions which have this feature enabled in the
COS allowed to them, will be given indication of the incoming call waiting on the trunk.
How it works
• A and B are extensions.
• CO Call Waiting feature is enabled in the Class of Service of B but not on A.
• There is an incoming call on a trunk for B.
• B is busy on a call with A.
• ETERNITY plays Beeps to B to indicate the call waiting.
• To answer the waiting call, B may dial Flash or press the HOLD key.
• B will be connected to the caller.
• A will be put on hold.
• When there is an incoming call on a trunk for A, but A is busy on another call, A will not be provided any
indication of the waiting call.
How to configure
For CO Call Waiting to work on an extension, it must be enabled in the Class of Service allowed to that extension.
This can be done using Jeeves as well as a Telephone.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. Template 01 is assigned CoS group 01. CO Call Waiting is disabled in the CoS. So, none of the
extensions of ETERNITY are provided call waiting indication for incoming trunk calls.
If you want to allow CO Call Waiting uniformly to all extensions of ETERNITY, simply enable this feature in the
default CoS group (01) in the default Template (01).
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the extensions to which CO Call Waiting is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for programming instructions.
What' this?
ETERNITY supports serial, asynchronous, RS232C Communication Ports.
ETERNITY PE6SP and PE3SP support a single communication port each. There is no communication port on
ETERNITY PE3SS.
A Communication Port is necessary for Programming ETERNITY using a PC, whereas for other above listed
facilities, Communication Port may or may not be used150.
How to configure
In order for each of the above listed facilities to work, a Communication Port must be assigned first as the
'Destination Port' and the attributes of the Communication Port of ETERNITY and the Communication Port of the
PC to which it is connected must programmed to match.
• PMS Interface
• “Configuring using Serial COM Port”
• “Station Message Detail Recording-Report”
• “Station Message Detail Recording-Online”
• “Station Message Detail Recording-Posting”
• “System Activity Log”
• “System Fault Log”
150. PMS can be interfaced on the Ethernet Port. For System Activity and Fault Logs, SMDR Reports and Online Printer Port can be
used. For SMDR-Posting Ethernet Port can be used.
• Speed in bps.
• Number of data bits.
• Number of stop bits.
• Parity
• Flow Control
• DSR Sensing
These attributes must be programmed keeping in mind the application for which the communication port is used
(for instance, Programming through PC, generating SMDR Reports, etc.)
When DSR Sensing is enabled, the system continuously monitors the physical connection between the
ports. Whenever the physical connection is detected to be inactive, the system stops data transfer. If DSR
Sensing is disabled, the physical connection between the ports is not monitored. Hence, even when the
physical connection is inactive, data transfer continues, and this data is lost.
The Communication Port attributes can be changed using Jeeves and dialing SE commands from a telephone.
• Speed (spd)
• Data Bits
• Parity
• Stop Bits
• Flow Control
• DSR Sensing
Speed is
151. Please note that maximum speed of the Communication port allowed in two-way communication like programming through com-
puter, programming through the Jeeves is 2400 bps only.
Parity is
0 for None
1 for Odd
2 for Even
3 for Mark
4 for Space
DSR Sensing
0 for Disable
1 for Enable
• Exit SE Mode.
3201-1-3
3202-1-1
3203-1-0
3204-1-0
3205-1-0
3206-1-1
3201-2-1
3202-2-0
How to use
For ETERNITY to communicate with a PC through the Communication Ports (COM1 and COM2), it must be
connected with the Communication Port of the PC.
You may connect any end to the ETERNITY and the other end to the PC.
If the PC supports only USB connectivity, use a USB-to-DB-9 converter of any standard make.
Refer the following Table for pin-out details of the COM Port.
5 Ground (GND)
152. This cable is supplied as an optional item. Contact your Matrix Dealer or the company to obtain this cable.
What’s this?
In a Dial-In, users can schedule a conference by dialing a conference number and a password and inform other
participants about the conference. The other participants can dial in to the on-going conference call to join the
conference.
Extension users can also schedule and join a Dial-In Conference from a remote location using “Direct Inward
Dialing (DID)”.
Conference Dial-In is useful for businesses to conduct client meetings or sales presentations, project meetings and
updates, regular team meetings, and communication to coworkers who operate in different locations. Besides
increased convenience, this feature allows workers to be more productive by saving time and cost of travel for out-
of-office meetings.
Number of
Maximum number of Parties in a
Model Simultaneous Dial-In
single Dial-In Conference
Conferences supported
ETERNITY ME 7 21
ETERNITY GE 5 15
ETERNITY PE 6S 5 15
How it works
A, B and C are extension users. C is at a remote location.
D, E and F are external callers.
• A dials the code for scheduling the conference, the conference number and the password.
The Conference Number must correspond with the number of simultaneous Dial-In Conferences
supported by the model of ETERNITY in use. For example, ETERNITY ME supports 7 Dial-In
Conference, so the, Conference number to the dialed by its users would be from 1 to 7. Similarly, the
ETERNITY PE 6S supports 5 simultaneous Dial-In Conferences, so the Conference number to the
dialed by its users would be from 1 to 5. The ETERNITY PE 3S/SP supports only 2 simultaneous Dial-
In conferences, so the Conference Number to be dialed by its users should be either 1 or 2. If a user
dials a conference number other than this, system will play an Error Tone.
• A informs B, C, D, E and F about the conference and provides them conference number, for example, '1'
and the password, '4040'.
• If C wants to schedule a conference, C must log into his extension from DISA mode.
• At the scheduled time, 4.30 pm, A initiates the conference by dialing the conference number and the
password, for instance: '1' and '4040'.
• When a new party joins the conference, the system plays beeps to the existing users, to inform them of
the new inclusion.
• Beeps are programmable. You can disable or enable beeps during conference.
Any extension user in a Dial-In Conference can include any other extension user or external party in the conference
by dialing the Conference Include feature code.
• If the conference has been initiated from the DISA mode, the caller (remote user) must dial the code for
withdrawing from the conference.
• If all participants of a Dial-In Conference have withdrawn from the conference, one-by-one, but none of
them have dialed the feature command to terminate the conference (190), the system will start the
'Release Conference if idle for more than (Minutes) Timer'. This Timer is programmable, and by default
it is set to 002 Minutes. On the expiry of this Timer, the system will free the resource occupied by the
conference on the conferencing circuit. However, any participant can join the conference before the
Conference Release Timer expires.
Canceling a Conference
If any of the extension users A, B or C wants to cancel the conference, they can do so by dialing the appropriate
feature code 190, the Conference Number and the Conference Password.
When an extension user cancels a conference, the system disconnects all the participants from the
conference, but does not free the resource occupied by the conference. To free the resource occupied by
the conference, you must release the conference (see Releasing a Dial-In Conference).
A Dial-In Conference can be canceled from SA mode also. This is useful when the participants forget the
password and cannot join the conference or cannot initiate the conference.
How to configure
To provide this feature to extensions,
• You must enable the feature 'Conference' in the“Class of Service (COS)” of the extensions in their “Station
Basic Feature Template”. By default, this feature is enabled on all extensions, so all extensions can use
this feature.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3-
Party and Multi-party Conference.
• If desired, you may also change default value of the Release Conference if Idle for more than (min.) Timer.
See “System Timers and Counts”.
• If extension users are to be allowed to initiate or join the Conference from a remote location, “Direct Inward
System Access (DISA)” must be enabled on the trunk on which they call.
You can also release a Dial-In Conference from System Administrator (SA) Mode using Jeeves. To do this,
• Open Jeeves.
• Enter the conference number (1 to 7) which you want to cancel in the Cancel Dial-In Conference Number
field.
When you enter DISA mode, you get beeps, dial digits before the DISA Inactivity Timer elapses.
Never dial 'Flash' when in DISA mode, you will get disconnected.
To cancel a conference:
• While you are all in speech, go ON-Hook.
• Go OFF-Hook.
• Dial 190.
What's this?
ETERNITY offers three types of conference calls: Conference-3 Party, “Conference Dial-In”, and “Conference-
Multiparty”.
Conference-3 Party (also referred to as Three-Way Calling) is a telephone call, in which the calling party can have
two other persons participate in the call.
A 3-Party Conference is initiated by dialing the number of the first person one wishes to talk to. The first person is
informed about the conference and put on hold. The number of the second person one wishes to talk to is dialed.
When the second person answers, s/he is informed about the conference. Three-way speech is established by
pressing Flash-0.
An already connected two-way speech can be converted into a conference by adding a second person, without
disconnecting the call with the first person.
Thus, a 3-Party Conference may be planned or conducted on the spur of the moment.
A 3-Party Conference can be conducted with extensions of ETERNITY and between extensions and external
numbers.
It is also possible to conduct an Unsupervised 3-Party Conference, wherein the operator connects two trunks
through the system and withdraws from the three-way speech.
The maximum number of simultaneous 3-Party Conferences supported by each model/variant of ETERNITY are
mentioned in the table below.
ETERNITY ME 7
ETERNITY GE 5
ETERNITY PE6S 5
How it works
A, B, C are extensions.
D and E are external numbers.
How to configure
For this feature to work, the feature 'Conference' must be enabled in the Class of Service group of the extensions
that are to be allowed this feature.
If extension users at remote locations are to be allowed to initiate the 3-party conference, “Direct Inward System
Access (DISA)” or “Direct Inward Dialing (DID)” must be enabled on the trunk on which their call lands.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
The feature 'Conference' in the Class of Service also includes Dial-In and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed all three types of
conferences - 3-Party, Dial-In and Multi-party Conference.
How to use
If Party 2 is a trunk,
• Dial #
• Dial 0 to grab a trunk. You get Trunk dial tone.
• Dial telephone number of Party 2. You get ring back tone.
• Speech with Party 2.
Now,
• Press DSS key assigned to 'Conference'.
OR
• Dial #
• Dial 0 to grab a trunk. You get Trunk dial tone.
• Dial telephone number of Party 2. You get ring back tone.
• Speech with Party 2.
• Dial Flash-0
• Three-way speech is established.
What’s this?
Like the Dial-In Conference, a Multi-party conference allows speech between more than three participants.
The key difference between Dial-In and Multi-party conference is that in a Dial-In conference participants can
include themselves in the conference by dialing into it without assistance, whereas in a Multi-party Conference the
party initiating the conference must include the participants by dialing their numbers and the Multi-party Conference
feature code.
Extension users can initiate multiparty conference from a remote location using “Direct Inward System Access
(DISA)”.
There are 7 digital conferencing circuits in the system. The ETERNITY supports between 6 to 21 parties in a
conference depending on the model you are using.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How it works
A, B, C, and D are extension users.
E and F are external numbers.
• Having included the last participant, F, A dials conference inclusion code again.
• A is included in the conference.
• All participants can now converse.
If the conference has been initiated from the DISA mode, the caller (remote user) must dial the code for
withdrawing from the conference.
If participant goes ON-Hook, without dialing the feature code for withdrawing from a Multi-party
Conference, the participant will be included in the conference again, when the participant dials the
Conference inclusion feature code (191). To avoid this, participants must dial the feature code for
Withdrawing from conference.
If all participants of a Multi-party Conference have withdrawn from the conference, one-by-one, but the
none of them have dialed the feature command to terminate the conference (190), the system will start the
'Release Conference if idle for more than (Minutes) Timer'. This Timer is programmable, and by default it is
set to 002 Minutes. On the expiry of this Timer, the system will free the resource occupied by the
conference on the conferencing circuit.
Canceling a Conference
Any participant in a conference can dial the Cancel conference code (190) to end the conference, all participants
will get Error Tone and the system resource occupied by the conference will be freed.
S1 A
S2 B
PSTN ETERNITY S3 C
S4 D
S5 E
S1 A
S2 B
S3 C
PSTN S4 D
S5 E
S6 W
ETERNITY
S7 X
S8 Y
S9 Z
S10 F
S11 G
A, B, C, D and E are in Conference1
W, X, Y and Z are in Conference2 S12 H
F, G and H are in Conference3
ETERNITY
T1 S1 A
T2 S2 B
T3 S3 C
PSTN T4 S4 D
T5 S5 E
T6
S6 F
S7 G
S8 H
S9 I
T1 S1 A
T2 S2 B
PSTN T3
S3 C
T4
S4 D
T5
S5 E
ETERNITY
T1 S1 A
T2 S2 B
PSTN T3
S3 C
T4
S4 D
T5
S5 E
ETERNITY
Simultaneous Multiparty conference between a few trunks and stations and between stations.
T1 S1 A
T2 S2 B
T3 S3 C
PSTN T4
T5 S4 E
T6 S5 D
S6 F
S7 G
ETERNITY
How to configure
To provide this feature to extensions,
• You must enable the feature 'Conference' in the“Class of Service (COS)” of the extensions in their “Station
Basic Feature Template”. By default, this feature is enabled on all extensions, so all extensions can use
this feature.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3-
Party and Multi-party Conference.
• If desired, you may also change default value of the Release Conference if Idle for more than (min.) Timer.
See “System Timers and Counts”.
How to use
To use Multiparty Conference from DISA mode, you may see the instructions “For Extension Users at a Remote
Location” under “Conference Dial-In”.
What's this?
You may recall that “Access Codes” are dialed at different call phases. No two Access Codes must be the same in
the same call phase.
For example, the same access code cannot be used for two different features like Call Forward and Redial, since
both these features are invoked in the 'Dial' phase. Similarly, Station and Logical Group Codes too must be unique
and should not match with any of the features invoked in the 'Dial' phase.
However, ETERNITY allows overlaps within Feature Codes and “Flexible Numbers” (Station Codes). One Feature
Access Code can be a part of (subset) another code, for example, 4, 41, 412; Flexible Numbers of extensions can
be 201, 2011 etc.
So, when such overlapping access codes are dialed, the system matches the first digit. On finding more than one
Access code starting with the same digit, the system will not know how to interpret the instruction and act
accordingly.
Conflict Dialing feature resolves this confusion. When an access code that is a subset of any other access code is
dialed, the system waits for some time for the extension user to dial the next digit. If the user does not dial any digit
within that time, the system interprets it as the smaller Access Code, and invokes the associated feature.
The time for which the system waits for the next digit to be dialed before resolving the Access Codes is called
"Conflict Dialing Timer". This timer is set to 2 seconds and is programmable.
How it works
You may set,
• If A does not dial any other digit before the Timer elapses, the system interprets the code as '41' and
invokes the Alarm feature.
• If such access codes exist, the system again waits for the duration of the Conflict Dialing Timer for another
digit to be dialed.
• Thus, only when the conflict in the access codes is resolved will the system respond accordingly.
How to configure
The working of this feature is controlled by the Conflict Dialing Timer, which is set by default to 2 seconds and can
be changed as desired.
If the duration of the Conflict Dialing Timer is long, it may cause delay in the system's response to the
feature. If the duration is less, the system may misinterpret the access codes. Ensure that the value of the
Timer is programmed optimally (i.e. at least the default value).
• Exit SE mode.
What's this?
Conversation Recording allows extension users to record their talk with other extension users or external parties,
after or without informing the opposite party.
This feature can be used to record verbal agreements, important discussions, instructions, interviews, client
requirements, take or place orders, etc.
Extensions must have a mailbox assigned to them for recording conversations. So, either a VMS card must be
installed or Matrix CadencePro must be interfaced with the system for this feature to work.
• Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
How it works
A and B are extensions. Both are assigned a mailbox each.
C and D are external parties.
• A calls C.
• C answers the call.
• A dials the command for Conversation Recording in mid-speech.
• C is put on hold.
• The system sends a string of digits to the Voice Mail System to initiate Conversation Recording.
• A and C are in speech again.
• The conversation recording starts in A's mailbox. The system plays beeps, if Conversation Recording
Beeps are enabled.
• A or C disconnects the call.
• Conversation recording ends.
• A can listen to the recorded conversation by invoking the Voice mail feature.
The same is repeated when B calls A. As both have mailboxes assigned, both can record the conversation.
How to configure
The functioning of this feature is controlled by three parameters: 'Class of Service', 'Mailbox', and 'Conversation
Recording Beeps'. These parameters can be programmed using Jeeves or by dialing SE Commands from a
telephone.
In the default Station Basic Feature Template Number 01 is assigned to all the extensions of ETERNITY, CoS
group 01 is assigned as default. Conversation Recording is disabled in CoS group 01. So, none of the extensions
of the ETERNITY can record conversations.
If you want to allow this feature to all extensions, simply enable Conversation Recording in the default CoS group
01.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the extensions to which Conversation Recording is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
Mailbox
Extensions that are to be allowed Conversation Recording must also have a mailbox. Refer VMS Card System
Manual or CadencePro System Manual for more information and programming instructions.
• Exit SE mode.
How to use
OR
OR
153. This is the default Voicemail Feature Access Code. Verify with you System Engineer if this has been changed and use the new
code.
154. Only if the mailbox is password protected.
What's this?
Customer Name is the name of the organization/enterprise that has deployed ETERNITY. As the User, you can
enter the name of your company/organization in the system.
When Customer Name is assigned in the system, this name will appear as header on the various System Reports
generated and printed by the ETERNITY like SMDR Incoming, Outgoing and Internal Call Reports, T1E1PRI
Performance reports, Alarm Status reports, etc.
The Customer Name may consist of a maximum of 80 alphanumeric characters, including punctuation marks. So,
you can enter the organization's address along with the Customer Name.
How to configure
Customer Name can be programmed using Jeeves and dialing SE commands from a Telephone at the time of
installation, or any time thereafter. It can also be corrected or changed any time.
• Exit SE mode
• Use EON to assign Customer Name, as SLT does not support alphanumeric dialing.
• The method of entering Customer Name from EON is similar to typing text messages from the cell
phone.
What's this?
Certain features of the ETERNITY like Operator, Class of Service, Toll Control, Outgoing Trunk Bundle Access
Groups, Trunk Landing Group, Direct Inward Dialing (DID), Direct Inward System Access (DISA), Security Alarms,
etc, require stations and trunks to behave differently according to the working hours, non-working hours and break
hours, which are referred to as Time Zones.
These Time Zone-dependant features and facilities are operated automatically according to the Time Tables
programmed in the system. In a Time Table, the Time Zones - Working Hours, Non-Working Hours, Break Hours -
are defined for the entire week. Time Table is assigned to trunks, stations and other time-zone dependant features.
The system executes the Time-Zone dependent features and facilities automatically according to the Time Table.
Day/Night Mode allows you to manually change the Time Zone of the system at any point in time, by issuing a
command. For example, the office is to be closed on account of an unplanned holiday or emergency. So, the Time
Zones of all stations and trunks must be set to Non-working hours to route outgoing calls and land incoming calls
from/to the appropriate destination. You can set the ETERNITY to Night Mode until the office remains closed and
set it back to operate as per the Time Table, when work is resumed.
To cite another example, the office must work for extended hours. You can set the ETERNITY to Day Mode and set
it back to operate as per the Time Table.
When you set the system in Day/Night Mode, the system overrides the Time Tables assigned to Trunks, Stations
and Operator. According to the mode you selected, it applies Working Hours/Non-Working Hours to run all the
Time-Zone dependent features of the system.
When the system is set to Day Mode, it applies Working Hours as the Time Zone for all stations, trunks and time
zone dependant features and facilities. When the system is set to Night Mode, it applies Non-Working Hours as the
Time Zone on Time-Zone dependent features of the system.
Thus, Day/Night Mode forces the system to work in a particular Time Zone, until it is changed again, manually.
Day/Night mode can be set by the System Engineer (SE mode) as well as by the System Administrator (SA Mode).
It can be done using Jeeves or by dialing a command from a Telephone.
How to configure
• Exit SE mode.
• Dial 1072-018-Code
Where,
Code is from 1 to 3
1 is for Day Mode
2 is for Night Mode
3 is for Operate system as per Time Table.
• Exit SA mode.
If you are setting Day/Night Mode from a DKP using a DSS key, refer the LED indication in table below.
What's this?
Daylight Saving Time (DST) is the practice of advancing clocks so that afternoons have more daylight and
mornings have less. Typically clocks are adjusted forward one hour near the start of spring and are adjusted
backward in autumn.
Many countries of the world use DST, though the start and end dates of DST vary with location and year. Even
within countries, uniform DST may not be observed. For example the states of Arizona and Hawaii do not observe
DST. Certain countries may observe DST in certain years, for instance Guatemala, while in most countries of Asia
and Africa, and in certain countries of South America, DST is not observed at all.
When ETERNITY is installed in a country/region where DST is used, it is necessary to synchronize the Real Time
Clock of ETERNITY with the local time.
So, if you are installing ETERNITY in a country where DST is used, find out the DST convention currently in use in
that country, and adjust DST accordingly.
How it works
The forward and backward adjustment of clocks can be Scheduled or Manual.
• Scheduled DST Adjustment: The Real Time Clock of the ETERNITY is advanced and set backward
automatically according to the DST convention of the country/region where the ETERNITY is installed.
Scheduled DST Adjustment is useful in countries/regions where DST Time is fixed, such as in Europe,
USA and Canada, without yearly variations.
The table below gives describes the DST conventions followed in the different countries for which
ETERNITY will automatically adjust DST.
01 Last Sun MAR Last Sun OCT Austria, Poland, Russia, Spain
From 01:59 to 03:00 From 02:59 to 02:00
03 Second Sunday MAR First Sunday NOV Bahrain, Mexico, Turkey, United States
From 01:59 to 03:00 From 01:59 to 01:00
07 Last Sun MAR Last Sun OCT Denmark, Ireland, Portugal, United Kingdom
00:59 02:00 01:59 01:00
The DST Type is to be selected according to the country/region where the system is installed.
When DST Mode is set to 'Scheduled' and the DST Type is selected, the system will automatically adjust
DST at the preset dates and time for the country/region where the system is installed.
For example, if ETERNITY is installed Spain, the DST Type 01 applicable to this country should be
programmed as Scheduled DST. The system will automatically advance the clock on the last Sunday of
March at 01.59.03:00 am every year (the start date of DST) and set the clock backward on the last Sunday
of October at 02.59.02:00 am of the same year.
• Manual DST Adjustment: The Real Time Clock of the ETERNITY is advanced and set backward
manually according to the DST convention of the country/region where the ETERNITY is installed.
Manual DST Adjustment is to be used in regions/countries that have no fixed DST Convention and where
yearly variations in DST practices are likely.
1. The 'Day of Month' method, which specifies a day of the month DST will start or end. For example:
starting on the 2nd Sunday of March and ending on 1st Sunday of November.
2. The 'Date and Month' method, which specifies a date of the month that DST will start or end. For
example: starting on March 11 and ending on November 4.
DST is not applicable in certain regions/countries, like Asia and South America. In such cases, the DST
Mode is to be 'Disabled'.
How to configure
• Click 'Submit' at the bottom of the page to save your DST setting.
• If you do not find your region on this list, you are recommended to set DST Mode to 'Manual' and adjust
DST manually.
• Go to the option 'Forward Time Adjustments' to advance the time when DST starts.
• 'Day-Month Wise' to specify the day of the month DST will start.
OR
• 'Date-Month Wise' to specify the date of the month DST will start.
If you select 'Day-Month Wise' option, the 'Date-Month Wise' option will be disabled, and vice versa.
Day-Month Wise
• If you select the 'Day-Month Wise' option, you should now select the desired options in each of the
following:
• Ordinal number: Select the Ordinal number of the day of the month, i.e. the 1st, 2nd, 3rd, 4th, 5th day,
when DST begins.
• Day: Select the day of the month - Sunday, Monday, Tuesday, Wednesday, Thursday, Friday, Saturday
- when DST begins.
• Change Time From: Select the time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
Date-Month Wise
• If you select 'Date-Month Wise' option, you should now select the desired options in each of the following:
• Change Time From: The time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
• To: The time to which the DST is advanced. The time mode is 24 hours, with options from 00 to 23
hours and 00 to 59 minutes.
• Now, go to the option 'Backward Time Adjustments' to set the time back (i.e. end DST and begin standard
time).
• Follow the same steps described above (step no. 4 to 6) to set the day/date, month, hours and minutes
except, here you must set these parameters according to the time when DST ends.
• Click Submit at the bottom of the page to save your DST settings.
• When the DST of a particular country starts or ends on the Last Sunday or any other day, for example,
the last Tuesday, last Friday of the month, always set the Ordinal Number as '5th'.
• Wherever time adjustments are made at 00:00 hours, use the previous date and set DST start time (i.e.
"from" time) at 23:59 hrs.
Please note that the advance time will be greater than the current time.
155. The current time is the time that is presently followed by the system.
156. This is the time to which the Real Time Clock should be advanced.
Please note that the delay time will be less than the current time.
DST
Applicable in Countries
Type
04 Brazil
05 Canada
06 Chile
08 Finland
09 Iraq
10 Kyrgyzstan
11 Egypt
12 Lebanon
13 Namibia
14 New Zealand
15 Norway
16 Paraguay
17 Syria
18 Cuba
• Exit SE mode.
157. This is the time to which the Real Time Clock should be set back to.
What's this?
• The DDI Routing Table is a set of general features that define the complete logic of identifying the flexible
numbers and DDI equivalent numbers when there is an incoming or outgoing call on a BRI/T1E1PRI and
SIP trunk. The ETERNITY offers 224 such tables each of which can be programmed as per the
requirement.
How it works
The DDI Routing Table is made up of the following Parameters:
• DDI Routing Reference ID-This is the reference number acts as an identifier to the mapping logic
programmed in the DDI Routing Table. Any number of table can have the same reference number. A
Reference ID is assigned to both IC reference tables and OG reference tables of the trunks. For more
details on call resolving, please refer the topics “Direct Dialing-In (DDI)”, “IC Reference Table” and “OG
Reference Table”.
• Start DDI Number-This is the First DDI Number for the ISDN Installation Number (MSN).
• Total DDI Numbers-The total number of DDI numbers supported for an ISDN Installation Number.
Suppose an ISDN Trunk supports 200 DDI numbers, then the total DDI Number will be 200.
• DDI Number of Digit-The number of digits in a DDI number. Suppose 200 DDI numbers are supported on
an ISDN Trunk, then the Number of Digits for that Trunk should be programmed as 3. Suppose 10 DDI
numbers are supported on another ISDN Trunk, then the Number of Digits for that Trunk should be
programmed as 2.
• Port Type-This parameter is to be programmed as listed in the parameter value in the DDI Routing Table.
• Port Number-Number of port, range of which depends upon selected port type. Refer parameter value
given in the DDI Routing Table. This field is of no significance if port type is selected as 'Flexible Number'.
• Start DDI Flexible Number-If port type is 'Flexible Number' you can enter the station number of the first
DDI Number for the Index. Once this is programmed based on the start DDI number the rest of the flexible
numbers of stations to which DDI Number is assigned is calculated. 'Start Flexible Number' field is of
significance only if the port type is 'Flexible Number'.
How to configure
Code takes different values that vary from parameter to parameter. Please refer default table, which provides all the
values that can be assigned to various parameters.
01 02 03 04 05 06
Para. No./ DDI Start Total Start
Table ID DDI Number Port
Routing DDI DDI Port Type Flexible
of Digit Number
Ref. ID Numbers Numbers Number
Parameter Value:
04 BRI 01-32
05 T1E1PRI 1-8
06 E&M 001-128
08 AOP 1
25 Mobile 01-64
26 SIP 01-32
• When a call is placed on SLT/DKP port, the calling party number is displayed on the terminal.
• When the call is placed on BRI-NT or PRI-NT, the calling party number and the called party number both
are sent to the NT port. Doing so, the PBX connected to the NT port can resolve the DDI number and place
the call on the programmed station.
Relevant Topics:
1. “Direct Dialing-In (DDI)” 1148
2. “ISDN-BRI” 1263
3. “T1E1 Trunks” 1608
4. “IC Reference Table” 1252
5. “OG Reference Table” 1346
What’s this?
• ETERNITY allows the System Engineer to monitor the state of software ports and IO operations. This is
known as Debug.
How to configure
Port Meaning
0 None
1 COM 1
2 COM 2
3 Printer
4 Ethernet
Card IO 0002
PMS 0004
Error 0016
VMS 0128
ACB 0256
Maturity/AOC 0512
• If it is required to enable combination of the above debugs, use value = Total Sum of the Values of all
requires debugs.
• This command will be saved in the configuration and hence, ETERNITY will retain this in case of power
failure.
Example:
• To enable debug for 'VMS' and 'PMS' use values = 0004 + 0128 = 0132 and give command 2104 - 0132.
• To disable debug for all process use command: 2104-0000.
01 SLT
02 DKP
03 TWT
04 BRI
05 T1E1
06 E&M
10 DOP
28 MOBILE
Software port number range shall be the software port number of the selected port type.
Flag Meaning
0 Disable
1 Enable
Settings made using this command is not saved in the configuration and hence if the system gets restart, SE needs
to give this command again.
Code Meaning
Default is 000.
Code Meaning
Default is 000.
It is not possible to enable all three debugs at a time, only one of the three can be enabled at a time.
Code Meaning
Default is 000
Code Meaning
0 Disable
1 Enable
If the Slot No. and the Port Number are programmed as 99 the debug of all slots and ports is generated.
By default, debug is 'disable'.
Program the IP address of the computer on which syslog server is running. ETERNITY will send debug on this IP
Address.
Use following command to program Port number on which ETERNITY shall send debug to syslog server:
2179-Syslog Server's Listening Port
Where,
Listening port can be from 1024 - 65535.
Default = 514
Program the Port Number on which the syslog server will listen.
Parameters programmed for the syslog server are applicable only when 'Ethernet' is selected as the destination
port.
Code Meaning
Relevant Topics:
1. “Debug VoIP Ethernet Port” 1089
2. “Communication Ports” 1038
What's this?
• Matrix ETERNITY supports a feature by which SE can view the system debugs on the server over IP
network. This is done by 'Syslog Client' in the ETERNITY which supports multiple debug levels.
• Each debug message includes the MAC Address of syslog client who is sending debug messages in
'syslog' format.
How it works
• The debug can be enabled/disabled.
• If Debug flag is 'Disabled', syslog client will not send any debug to 'syslog server address'.
• If Debug flag is 'Enabled', syslog client will send debug to 'syslog server address' and the 'Debug Level'
which is enabled, can be viewed on the syslog server. The Server will use the MAC address data to filter
the debug of the specific gateway.
• The Server address and Server port number are used as programmed values.
• System
• Serial
• SIP
• CALL
• Registered User
• Registered Trunk
• BLF
• MWI
• Media
• VoPP
• Call Advance
• Registered User Advance
• Registered Trunk Advance
• BLF/MWI Advance
• Media Advance
• Presence
• IM
• As per the level selected, debug log will be generated. For example: if debug log of Call is required, enable
'CALL' level and disable all other debug levels.
Use following command to enable/disable Debug for the VoIP Ethernet Port:
7791-1-VoIP Ethernet Port -Debug
7791-2-VoIP Ethernet Port -VoIP Ethernet Port-Debug
7791-*-Debug
Where,
VoIP Ethernet Port is from 01 to 16.
Debug Meaning
0 Disable
1 Enable
By default: Disable.
Use following command to assign Syslog Server Address on which Debug parameter is to be sent for the VoIP
Ethernet Port:
7792-1-VoIP Ethernet Port-Syslog Server Address
7792-2-VoIP Ethernet Port-Syslog Server Address
7792-*-Syslog Server Address
Where,
VoIP Ethernet port is from 01 to 16.
Syslog Server Address is of 15 digits maximum.
By default, Blank.
Use following command to assign Server Port Address on which Debug parameter is to be sent for the VoIP
Ethernet Port:
7793-1-VoIP Ethernet Port-Server Port Address
7793-2-VoIP Ethernet Port-Server Port Address
7793-*-Server Port Address
Where,
VoIP Ethernet Port is from 01 to 16.
Server Port Address is from 1024 to 65535 and 514.
By default: 514.
If Debug flag is Enabled and Syslog Server IP address is Blank, debug log will be generated on Console
port i.e. COM port. The Baud rate on Hyper terminal should be set to 115200.
Use following command to enable debug level for the VoIP Ethernet port:
7794-1-VoIP Ethernet Port-Index-Code
7794-2-VoIP Ethernet Port-VoIP Ethernet Port-Index-Code
7794-*-Index-Code
Where,
VoIP Ethernet Port is from 01 to 16.
Index Meaning
01 System
02 Serial
03 SIP
04 Call
05 Registered User
06 Registered Trunk
07 BLF
08 MWI
09 Media
10 VoPP
11 Call Advance
14 BLF/MWI Advance
15 Media Advance
16 Presence
17 IM
Code Meaning
0 Disable
1 Enable
What's this?
ETERNITY is supplied with preset values for system and feature settings, as which may be altered and customized
by users to match their requirements and preferences. The factory-set values for system and feature settings that
are automatically assigned by the system are referred to as Default Settings or standard settings.
Every configurable parameter in the system has factory-set default values, which may be changed or customized to
match user requirements and preferences.
How it works
The default settings are to be loaded or restored in the following situations:
So, you must select the appropriate “Region” for the country/region in which the system is installed.
The system will load the default settings for the country/geographical region where the system is installed.
The system is designed to work efficiently with the default settings. So, if the country/region-specific
default settings match their requirements, you may not even need to alter or customize the values of
various parameters,
They may work with default settings for the most part, customizing only some of the parameters to match
their specific requirements.
The country-specific default settings of various parameters that will be loaded on changing the 'Region' are
presented in the table below. For default values of Trunk Access Codes, Emergency Numbers, Distinctive
Rings, for various countries refer the respective topics.
Default
Default Default Default
Country Country DST Distinctive Abbr.
Time DST CPTG DKP Opr TAC
Code Name Schedule Ring Dialing
Zone Mode Language
Type
001 Afghanistan GMT+04:30 English
002 Algeria GMT+01:00 English
003 Antigua and GMT-04:00 English
Barbuda
004 Argentina GMT-03:00 04 Spanish
005 Australia GMT+08:00 Scheduled 2 05 English 9
(Perth)
006 Australia GMT+09:30 Scheduled 2 05 English 9
(Note2)
(Adelaide)
In addition to the common set of PBX features, there is a distinct set of in-built features for each of these
applications. When ETERNITY is to be installed in any of the two application scenarios, the 'Customer
Profile' - whether the user is an Enterprise or a Hotel - is to be defined at the time of installation.
When the Customer Profile is defined, all features specific to the application Enterprise/Hotel, along with
their default settings are loaded. By default the Customer Profile of ETERNITY is defined as 'Enterprise'.
When there is a system malfunction, possibly caused by a programming error that you are unable to
diagnose, you may restore default settings.
To be able to do this, you must have the Programming Password, also referred to as the "SE password".
Whenever you restore the default settings in the system, all the programmable parameters except Network
Port Parameters158 and the “Region” will be set back to their default values.
158. The IP Address, Subnet Mask, Primary DNS, Secondary DNS, Host Name, Domain Name, DHCP Server Address.
• The SE password you enter must be the current password. For Example: if it is 1234, enter 4321 and click
OK.
For Example: if your current SE password is 1234, you must dial 5302-4321 to restore default settings.
• The names programmed for the DKP and SLT will disappear and the default flexible numbers, i.e.
extension numbers assigned to the DKP and SLT stations will appear.
Without the SE Password, you cannot restore default values via the software. If you forget the SE
Password, you must resort to hardware default of the SE-Programming Password first. Refer the topic “”
later in this section for instructions on restoring the default SE Password.
You cannot default “Region”; you can only select a region to load the country-specific default settings and
default the system.
What's this?
Department Call enables you to group together extensions of a particular department so that callers can reach
anyone in the department by dialing a common access code assigned to the department.
Calls made to such groups of extensions are called Department Calls and the access code used to make
department calls is called Department Number.
This feature is useful in situations where any member of a department may interact with callers, as for instance in a
information counter, a customer care cell, a technical support team, etc.
Callers can also reach individual extensions in a Department group by dialing the extension number.
ETERNITY supports the formation of 16 department groups. The member extensions of a department group may
be single line telephones (SLT), digital key phones (DKP), SIP Extensions, ISDN Terminals.
How it works
Extensions A, B, C, D are grouped as a Department with the access code 3901.
Internal Calls
• Extension E dials 3901 to call the Department.
• The system checks E's Class of Service for the Department Call feature.
• The feature is enabled. The system checks if Rotation is enabled in the routing group assigned to the
Department.
• The Rotation flag is enabled. The system lands the call on the extension which is set to ring first.
• Extension A, configured as the first landing destination rings for the duration of the Ring Timer
(configurable; default: 15 seconds).
• A answers the call. Speech established between A and E.
• If A does not answer, the system hunts for the next extension in the group to land the call, B.
• B starts ringing for the duration of the Ring Timer.
• If Continuous Ring is enabled on A, A will continue to ring even as B is ringing.
• If B does not answer the call at the end of the timer, the system hunts for the next extension, C.
• If B has Continuous Ring enabled, B will continue to ring even as C is ringing.
• If the call is not answered even after hunting the last extension, the system will loop back and start from
the first extension once again.
External Calls
Department Calls can be made using DID. For example, a company may use DID to have callers who want
information only to dial the Information Department instead of waiting for the Operator.
Thus for each call, the system will hunt for a landing extension as per the Rotation set for the routing group. The
extensions will ring for the duration of the Ring Timer, either continuously or one-by-one (as per the Continuous
Flag configured), and according to the sequence in which the extensions in the group are arranged.
Rotation ensures equal distribution of call traffic. If Rotation is disabled, the fresh call will always land on first
extension of the Department group.
Department Group and ISDN Terminal: When more than two ISDN terminals connected to the same BRI
port are configured as members of a Department group, if a call is made to this group using the department
group access code, only two ISDN terminals connected to the BRI port will ring. This limitation is because
of the BRI protocol.
How to configure
The functioning of this feature requires you to do the following:
• Create Department Groups.
• Configure “Routing Group” (each routing group consisting of extensions related to a Department) and
assign the routing groups and appropriate access codes to the department groups.
• Enable 'Department Call' in the Class of Service (COS) of extensions that are to be allowed to make
Department calls.
• Decide the number of department groups you want to create, for example: 4 groups.
• Group all the extensions you want to put in each department group. You cannot group more than 32
extensions in a single department group.
• Decide in what sequence the extensions in each group should ring, i.e. which extensions should ring first,
second, third, and so forth.
• Decide the access code you want to assign to each department group.
3010, 3011,
1 3901 2001, 2002, 2003 3301, 3302
3012
The access codes for the department groups and extensions in this table are default access codes.
• Now, with this information ready, you may configure the department groups using Jeeves or dialing the
relevant SE commands from a Telephone.
In the default Station Basic Feature Template Number 01 is assigned to all the extensions of ETERNITY, the
default COS group 01 in the template has 'Department Call' included in the ‘Basic Features’. So all extensions can
make Department calls. You cannot deny this feature to any extension, without denying the entire set of Basic
Features.
If you wish to allow this feature to member extensions only, retain this feature in the COS group of member
extensions only, and disable this feature in the COS group of all other extensions. For this, you may create
separate templates for member extensions and other extensions.
Refer the topic “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
By default, the Access Codes assigned to Department groups are from 3901 to 3916.
If you decide not to use the default access codes, ensure that the access code you assign to each
department group is unique and does not match with any SLT, DKP, ISDN, SIP Extension, DOP access
code or any feature access code of the Dial Phase. Refer the topic “Access Codes” to know more.
• Now, enter the Routing Group, i.e. the number of the group you created for this department.
Where multiple departments exist, you must create separate routing groups for each department group.
• create the routing groups first and simply enter the relevant routing group number against the
Department Group Index (to which you have assigned the access code).
OR
• Choose the Routing Group number (from 01 to 96) you want to assign to the department group. In each
routing group you can include a maximum 32 extensions as 'members'.
• For each routing group you want to assign to a department group, configure the following parameters:
• Rotation Flag: With this flag, you can enable or disable the rotation of calls in the routing group
which has multiple 'member' extensions. When enabled, each fresh call will land on the extension
which is next to the one that received the last call. This ensures equal distribution of incoming calls
to all the destinations within the routing group. The flag has no relevance if the routing group has
only one member extension.
• Member Type: Select the 'Member Type' from this list. If the extension you want to add as member
of the group is an SLT, select SLT; if it is a DKP, select DKP as member type. Similarly, select ISDN
if the extension is an ISDN Terminal and SIP Extn. if the extension is a SIP extension. Include only
as many extensions as you want in the routing group and set the remaining Member Types to
'None'. For example: if you want to include only two extensions in the routing group, set the Member
Type in the remaining columns (Member 03-Member 32) to 'None.'
• Port Number: Enter the software port number on which the SLT/DKP/SIP Extension/ISDN terminal
to be grouped is attached.
• Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should
ring. By default, the ring timer is set to 015 seconds and can be changed.
• Continuous Ring Flag: With this flag, you can set a extension to ring continuously until the call is
answered. The first extension will continue to ring even as the system hunts for other extensions in
the routing group to land the call. If the call still remains unanswered, the system will return the call
• Repeat the above steps to include other extensions in the routing group.
For example:
The Customer Care Department of a company has four extensions: 201, 202, 203 and 204 (on software ports 001,
002, 003, and 004 respectively), which needs to be grouped for Department Calls.
Extensions 201 and 202 are SLTs. Extensions 203 and 204 are DKPs.
Solution: Select a routing group, for example 03, and configure as follows:
1. 201 as member 01, with member type SLT, and Port number 001.
2. 202 as member 02, with member type SLT, and Port number 002.
3. 203 as member 03, with member type DKP, and Port number 003
4. 204 as member 04, with member type DKP, and Port number 004.
1. Enable the 'Rotation Flag' on routing group number 03 to distribute call traffic.
2. Enable the 'Continuous Ring Flag' for member 01 (201) and set the 'Ring Timer' to '20 seconds.
3. Set the Ring Timer of member 02 (202) to 10 seconds. Disable 'Continuous Ring' flag.
4. Retain the Ring Timer of member 03 (203) as default 15 seconds. Disable 'Continuous Ring' flag.
5. Set the Ring Timer of member 04 (204) to 20 seconds. Disable 'Continuous Ring' flag.
Port Number is the Software port number on which the member extension SLT, DKP, SIP Extension,
ISDN Terminal is attached.
Software port number of the SLT, from 001 to 512.
Software port number of the DKP, from 001 to 128.
Software port number of the ISDN Terminal, from 01 to 64.
Software port number of the SIP extension, from 001 to 999.
To set the Ring Timer for each member extension in the routing group, dial:
• 6503-1-Routing Group-Destination Index-Ring Timer
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is number of the member extension in the routing group from 01 to 32.
Ring Timer is from 000 to 255 seconds. (Default: 015 seconds)
For Example: To set the Ring Timer for the individual extensions of the Routing Group 03 in the above
example, dial:
To set the Continuous Ring Flag for extensions in the routing group, dial:
• 6504-1-Routing Group-Destination Index-Flag
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is the number of the member extension in the routing group from 01 to 32.
1 for enable continuous ring (the first extension in the group will ring till the call is answered).
For Example: To enable/disable the Continuous Ring Flag for the individual extensions of the Routing
Group 03 in the above example, dial:
For Example: To assign Routing Group 03 to Department Group 01 (Customer Care) as in the above
example, dial 2001-1-01-03.
For Example: To assign '51' as Access Code for the Department Group 01 as in the above example,
dial 3113-1-01-51-#*
• Exit SE mode.
OR
What's this?
Dial By Name enables extension users to call another extension or an external party by dialing the name of the
person, instead of dialing their telephone number.
This feature is accessible only to users of the proprietary digital key phones and the Extended IP phones of Matrix.
With Dial By Name users need not remember the desired party's telephone number or short codes, i.e.
“Abbreviated Dialing” codes.
For each extension, the database for names used in Dial by Name is drawn from:
• the Personal Directory, which is assigned to each extension, wherein up to 25 external party numbers
along with their names may stored. The system uses the Personal Directory to dial external parties by their
names. See “Abbreviated Dialing” to know more.
• Global Directory, which is assigned to the extension in its “Class of Service (COS)”. The Global Directory
is a system-wide list of external party numbers and names. Up to 999 numbers can be stored in this
directory, and parts of the Global Directory (Part 1, 2, 3) can be assigned to each extension in its Class of
Service. See “Abbreviated Dialing” to know more.
• Names of Extensions, which are names of users/departments groups. Their names are assigned to SLT,
DKP and SIP extensions to identify the extension users. Names of Extensions are necessary for making
internal calls using the Dial By Name feature.
How it works
• Extension user presses the DSS Key assigned to 'Dial By Name' feature.
• On EON48 model and on SETU VP248, press the 'Names' key.
• On EON42 model, press the 'Dial by Name' key
• The prompt <Name: > appears on the phone display.
• User enters the name of the desired party159.
• For example, user wants to call Midas Biz, and enters the letter 'M' using the keypad.
• The system displays in alphabetical order, all names starting with 'M'. These numbers are drawn from the
Personal and Global Directories assigned to the extension and the Extension Names programmed in the
system.
• User scrolls the list using the Up/Down navigation keys to reach the desired contact's name.
OR
Instead of scrolling the entire list, the user enters more than one initial letter of the contact's name. The
search is narrowed down to more accurate matches. The phone displays the matching entries in the
directory.
• The user must select the desired name by pressing 'Enter' Key.
159. The process of entering the names is the same as when writing text messages (SMS) from a cell phone. The keys must be
pressed multiple times in quick succession to enter the desired alphabet.
How to configure
For this feature to work, the following must programmed:
1. DSS Key: A direct station selection (DSS) key must be programmed for the Dial by Name feature. Without
the DSS Key this feature will not be accessible.
The factory-default key map of EON48 and SETU VP248, both phones have the DSS Key labeled as
'Names'.
2. Global Directory: The names of the external parties must be programmed against their respective
telephone numbers in the directory. Refer the topic “Abbreviated Dialing” for instructions on programming
the Global Directories.
3. Personal Directory: The names of the external parties must be programmed against their respective
telephone numbers in the Personal Directory. Refer the topic “Abbreviated Dialing” for instructions on
programming the Personal Directories.
4. Extension Names: Extensions may be SLTs, DKPs, ISDN Terminals or SIP extensions. Refer the topics
related to configuration of extensions160.
5. Class of Service: Dial By Name is allowed to all DKP users. However, the use of this feature is related to
the following features, which must be enabled in the Class of Service of the DKP and SIP extension users:
• 'Internal Calls'- This is to be enabled so that the station can call other stations.
• Global Directory Part 1
• Global Directory Part 2
• Global Directory Part 3.
Global Directory Part 1 is assigned to the default CoS group 01 assigned to all stations in the default
Station Basic Feature Template 01.
If you want the names to be drawn from Global Directory Part 2 and Part 3, provided these are
programmed, you must enable these two directories in the CoS of the DKP and SIP extensions.
Refer “Class of Service (COS)” and “Station Basic Feature Template” for programming instructions on how
to enable a feature in the CoS and how to apply it on extensions.
The system will display the names exactly as they have been programmed in the Personal and Global
Directories and the SLT/DKP/ISDN Parameters. Refer the topic “Configuring DKP Extensions”.
160. You may also refer the instructions provided under the topic Configuring Extensions: “Configuring SLT Extensions”, “Configuring
DKP Extensions”, “Configuring ISDN Terminals”.
• Go ON-Hook.
• Go OFF-Hook.
• Press the DSS Key labeled 'Names'/'Dial By Name' again.
• Enter the name/initial letters of the contact's name.
What's this?
Dialed Number Directory is a Digital Key Phone feature, available only to the users of the proprietary DKP, Eon.
It is the list of numbers dialed out from the DKP, similar to the call history of recently dialed calls on a cell phone.
These numbers may have been dialed out using features like Abbreviated Dialing, Quick Dial, Redial, Walk-In
Class of Service, or may be a simple outgoing call made by directly dialing the external number.
How it works
• When a DKP extension user makes an outgoing external call, the number is stored in the Redial Number
List.
• The list is updated using the First-In First-Out logic, whereby the earliest dialed number is replaced with
the most recently dialed number.
• To use this feature, the DKP user must invoke the “Last Number Redial” feature.
• Doing so, the Redial Number List will appear on the phone display.
• The user may now navigate the list, select the number to be dialed out.
• The system will dial out the selected number using the same Outgoing Trunk Bundle Group used to place
this call earlier.
• If the number had been dialed earlier using Abbreviated Dialing, the system will check for Toll Control
when dialing out the number again from the dialed number directory161.
How to configure
No specific programming required.
How to use
OR
161. Recall that the system does not check for Toll Control when Abbreviated Dialing is used.
What's this?
Digest Authentication is a challenge-based authentication service of SIP to authenticate the identity of the
originator of SIP request in the INVITE message. The recipient of the request can ascertain whether or not the
originator of the request is authorised to make the request. When the digest credentials of the originator—User
Name and Password—in the INVITE message are authenticated and accepted by the recipient, the originator and
the recipient are connected.
How it works
The Digest Authentication feature works on the basis of the Digest Authentication Table, in which the credentials,
namely the User Name and Passwords of trusted/authorised calling party SIP devices are stored. You must
configure this table. The Digest Authentication Table is common for all SIP trunks on which this feature is enabled.
When you enable this feature on a SIP trunk, for all incoming calls (SIP requests),
• ETERNITY will challenge the identity of the calling party (the SIP device initiating the request) to send its
digest credentials.
• When the calling party sends its credentials, ETERNITY authenticates the credentials by matching it with
its Digest Authentication Table.
• If a match is found, the calling party will be authenticated and the call will be allowed on the SIP trunk.
• If no match is found, ETERNITY will consider it as invalid authentication information and reject the call.
How to configure
To use this feature on SIP Trunks, you must do the following:
• Enable Digest Authentication on the SIP trunks you want to use this feature.
• Configure the Digest Authentication Table.
You can configure the Digest Authentication Table using Jeeves and from a telephone.
• In the Password field, enter the corresponding Password. The Password must be within 16 characters.
• Now, enable Digest Authentication on the desired SIP trunks. For instructions, see “Configuring SIP
Trunks”.
• Exit SE mode.
For SE Command for enabling Digest Authentication on SIP trunks, see “Configuring SIP Trunks”.
What's this?
The ETERNITY provides a single reliable, solid-state Digital Input Port (DIP) on all models of ETERNITY. Any
external sensor device or panic switch like an object sensor, smoke sensor, glass-break detector, water level
sensor, can be connected to the DIP.
The DIP is located on “The Master Card” of the ETERNITY ME, on “The CPU Card” of the ETERNITY GE, and on
“The Door Phone Card” in the ETERNITY PE.
The DIP can be used to trigger “Automated Control Applications” wherein, a gadget connected to the Digital Output
Port can be turned ON and OFF on instigation from the DIP.
A typical example of an automated control application would be of a fire alarm or a smoke sensor connected to the
DIP. Whenever it senses smoke, it sends an instigation to switch ON the hooter or siren connected to the Digital
Output Port.
Another example would be an object sensor for turning on lights on the premises. This sensor connected to the DIP
sends instigation to the Digital Output Port, whenever it senses the presence of persons or objects on the premises.
The Digital Output Port turns on the light on receiving this instigation from the DIP.
The DIP can be used for triggering Security Alarms, by connecting a fire/smoke sensor, a break-in or glass-break
sensor to it.
How it works
• A sensor is connected to the DIP.
• The system waits for the duration of the Minimum Instigation Time (programmable; default: 01 second).
This is the time for which the instigation signal from the sensor should remain present on the DIP for it to
be identified as a genuine signal.
• If the DIP is being used for an “Automated Control Applications”, the system will instigate the “Digital
Output Port (DOP)” which will turn ON or OFF the Digital Output Port and hence the gadget connected
to it on receiving the instigation.
• If the DIP is being used for Security Alarm and Reporting, the system will trigger the alarm device -
hooter/siren - connected to the Digital Output Port or it will make a call to the external number or the
Routing Group programmed as destination for triggering Security Alarms.
• It will also ‘report’ the alarm call to the group of extensions programmed to receive Security Reporting
calls.
Refer the chapter “Installing ETERNITY” for step-by-step instructions on connecting a sensor device to the DIP of
your model of ETERNITY.
Do not connect devices that do not conform to the specifications of the DIP!
How to configure
Whether you are using the DIP for an automated control application or for Security Alarm and Reporting,
you must configure the following parameters:
• 'High' state signifies that the DIP is normally open. DIP should be programmed as 'High' when the
sensor connected to the DIP keeps the Loop open and closes it to signal an event.
• 'Low' state signifies that the DIP is normally closed. DIP should be programmed as 'Low' when the
sensor connected to the DIP normally keeps the Loop closed and opens/breaks it to signal an
event.
• Minimum Instigation Time: This is the time for which the instigation signal from the sensor device
should remain present on the DIP to be recognized by the DIP as a genuine signal. The range of this
timer is from 01 to 99 seconds. By default the Minimum Instigation Time is set to 01 second. You may
set the 'Minimum Instigation Time' to the desired value.
• Exit SE Mode.
DKP Features
• Status of other ports (Tri-color LED indication)
• Programmable Direct Station Selection (DSS) Keys and Feature keys
• LCD notification messages
• Ringer Tune selection
• Adjustable Speech level
• Adjustable Ringer Volume
• Adjustable Backlight and Contrast levels
• Hands-free operation - Speaker key and headset connectivity.
• Call Logs - last 20 Missed, Answered and Dialed Calls.
• Operator, Executive, Hotel Attendant and Guest Functionality
• Message Paging
• Menu based operation of PBX features
• Multiple Language support.
PBX Features
Listed below are the features of ETERNITY that require a Digital Key Phone:
• Abbreviated Dialing
• Auto Answer
• Background Music
• Call Chaining
• Call Cost Display
• Call Duration Display
• Call Mute
• Dialed Number Directory
• Directory Dialing by Name
• Dynamic Lock
• Forced Answer
• Keypad Lock
• Live Call Screening
• Message Paging
• Off-Hook Alert
• Room Monitor
• Text Message Reply
• Time Zone Display
• User Status (Presence)
Model
Feature
EON42 EON48S EON48P EON48DS EON48DP
Total number of keys 42 48 48 48 48
Number of programmable keys 25 29 29 29 29
Capsense keys No Yes Yes Yes Yes
LCD display capacity 2 lines x 24 2 lines x 24 6 lines x 24 2 lines x 24 6 lines x 24
characters characters characters characters characters
Touch Keys No No No No No
Touch screen operation No No No No No
LCD with backlight Yes Yes Yes Yes Yes
Headset Interface Yes Yes Yes Yes Yes
Ringer Lamp (LED) Yes Yes Yes Yes Yes
Speaker Phone Full duplex Full duplex Full duplex Full duplex Full duplex
EON 42
2 lines and 24 characters LCD display, full duplex speaker phone with headset connectivity.
LCD Display
The LCD display of EON42 is backlit and is similar to the one on a folder-type mobile phone. You can tilt the display
at a convenient angle for a clear view of the text/characters displayed.
The backlight cannot be switched ON or OFF by opening or closing the LCD display. The LCD backlight can be
turned on and off as well as adjusted for contrast and brightness from the "Phone Settings" of the DKP Phone
Menu.
The Ringer LED changes color according to the type of call, as described in the table below.
Priority Red
Navigation Keys
• Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you press the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
The Forward Key can be used to increase the volume of Speech when talking to someone and to increase
the volume of the Ringer when the phone rings.
• Back Key: To move backwards when dialing a number; to go back one level in the Menu.
The Back Key can be used to decrease the volume of Speech when talking to someone and to decrease
the volume of the Ringer when the phone rings.
Thus the status of the DKP user's own Station as well as that of the other Stations and the status of Trunk
lines are indicated by the LED of the DSS keys assigned to those Stations and Trunks on the DKP.
The following table shows the relationship between the color of the LED and various events:
Green The key assigned to the The key assigned to the The key assigned to the Station
Station you are in speech Station. you are calling or from which you
with. you have kept on hold. are being called.
Red The key assigned to the The key assigned to the The key assigned to the Station/
Station that is now busy Station which has put Trunk that is called or being
with another Station/ another Station/Trunk on called by another.
Trunk. hold.
Orange You are talking on a Trunk You have held a Trunk You have an incoming call on the
(external call) (external call) Trunk (external call)
• Green indicates the state of the station/trunk you access. For example, when you make a call to
another Station 203, the LED of the DSS key assigned to Station 203 blinks Green to indicate ringing at
the Station. If you have successfully established speech with Station 203 the LED glows Green
continuously.
• Red indicates the state of other Stations/Trunks. For example, if the LED of the DSS key assigned to
Station 201 is glowing Red continuously, it means Station 201 is busy with another Station or Trunk.
• Orange indicates the state of the trunk you are in speech with. For example, when you are in speech
on an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
Green When you are in speech When you have put a When any Station is calling
with a Station (internal call) Station on hold (internal call) (internal call)
Orange When you are in speech When you have put a Trunk When any Trunk is calling
with Trunk (external call) on hold (external call) (external call)
• Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Stations/Trunks glow in a single color - Red - to indicate status of the
call event on the Stations/Trunks and on the DKP.
• For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
• A feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the
DSS key to which the Auto Redial feature has been assigned will glow Red, when Auto-Redial is set,
and the LED is turned off when the feature is canceled.
• Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive.
• The LEDs of DSS keys to which features like Raid, Interrupt Request, Barge-In, Last Caller Recall are
assigned, will not glow.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of stations, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key is the last key on bottom-right of the keypad. It sets the phone in 'Speaker mode' for hands-free
operation. The Speaker key is programmable, you can assign any other feature/function to this key.
Since key is programmable, the LED indication pattern will be according to the feature/function you assign to this
key. For example, if you assign a Station to this key, the LED of the key will function as a tri-color LED to show
status of the Station. If you assign a feature that does not require any LED activity, like Call Pick-Up, the LED of this
key will remain inactive.
Headset Connectivity
The EON42 provides a Headset interface. You can use any stereo headset of standard make with a 2.5 mm single
connector into the headset jack on the left side panel of the phone.
You can also assign Headset key function to any of the DSS keys. Refer the topic “DSS Keys Programming” for
instructions.
Key Maps
EON may be the extension of the Operators and Executives in an enterprise, and the extension of the Front Desk
Attendant and Guest in hotels. Some of the feature required by each of these groups may not be required by
others.
For example, when EON is an Operator's extensions, more DSS keys would be required for Trunk Access, Call
Appearances, and Direct Station Calling, than for features. But when EON is a Hotel Attendant's extension, keys
are required for specific hotel functions such as Check-In/Check-Out, Changing Room Clean Status, Room Shift,
etc.
1 2 3 1 2 3 1 2 3
4 5 6 4 5 6 4 5 6
7 8 9 7 8 9 7 8 9
* 0 # * 0 # * 0 #
These key maps can be customized to match the exact requirement of individual users. Refer the topic “DSS Keys
Programming” for instructions on customizing the Key Maps.
Phone Menu
You can access the following PBX and phone features from the Menu of EON42:
Call Logs To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Contacts To add, edit, delete names and numbers of contacts in the Global Directory Part 1.
Call Forward To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-Forward-
Unconditional, and Follow Me.
Do Not Disturb To set/cancel Do Not Disturb on the phone, i.e. block incoming internal and external
calls.
Call Cost Display To view the cost of calls made from the phone.
Change User To change User Password (required for using certain features like Call Follow Me,
Password Dynamic Lock, DISA, Walk-In Class of Service, User Absent/Present, Hot Desk, Voice
mail) and for customizing Phone Settings.
Phone Settings To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).
To exit menu,
• Go ON-Hook
Or
If you go ON-Hook, the call waiting beeps will be turned into an audible Ring indicating an incoming call.
Operating EON42
Please refer the User Card for EON42 for instructions on operating the features of ETERNITY using EON.
EON48S/EON48D-S
2 lines and 24 characters LCD display, full duplex, capsense feature keys
EON48P/EON48D-P
6 lines and 24 characters LCD display, full duplex, capsense feature keys.
LCD Display
The LCD display of EON48P/48D-P/48S/48D-S is backlit and can be tilted at a convenient angle for a clear view of
the text/characters displayed.
Ringer LED
The Ringer LED indicates incoming internal and external calls. The LED Cadence will match with the Ring
Cadence of the incoming internal/external call.
The Ringer LED changes color according to the type of call, as described in the table below.
Priority Red
Navigation Keys
These are 5 capsense keys. The functions of each are described briefly below.
• Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you tap the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
• Back Key: To move backwards when dialing a number; to go back one level in the Menu.
Refer the topic “DSS Keys Programming” for instructions on programming these keys.
• Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks glow in three colors to
indicate status of the call event on the Stations/Trunks and on the DKP.
The following table shows the relationship between the color of the LED and various events:
Blue The key assigned to the The key assigned to the The key assigned to the
Station you are in speech Station you have kept on Station you are calling or from
with. hold. which you are being called.
Red The key assigned to the The key assigned to the The key assigned to the
Station that is now busy with Station which has put another Station/Trunk that is called or
another Station/Trunk. Station/Trunk on hold. being called by another.
Violet You are talking on a Trunk You have held a Trunk You have an incoming call on
(external call) (external call) the Trunk (external call)
• Blue indicates the state of the station/trunk you access. For example, when you make a call to another
Station 203, the LED of the DSS key assigned to Station 203 blinks Blue to indicate ringing at the
Station. If you have successfully established speech with Station 203 the LED glows Blue continuously.
• Red indicates the state of other Stations/Trunks. For example, if the LED of the DSS key assigned to
Station 201 is glowing Red continuously, it means Station 201 is busy with another Station or Trunk.
• Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on
an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
Blue When you are in speech with When you have put a Station When any Station is calling
a Station (internal call) on hold (internal call) (internal call)
Violet When you are in speech with When you have put a Trunk When any Trunk is calling
Trunk (external call) on hold (external call) (external call)
• Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Stations/Trunks glow in a single color - Red - to indicate status of the
call event on the Stations/Trunks and on the DKP.
• Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
• For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
• Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive for example, Raid, Interrupt Request, Barge-In, Last Caller
Recall.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of stations, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can assign any other feature/function on this key.
Since key is programmable, the LED indication pattern will be according to the feature/function you assign to this
key. For example, if you assign a Station to this key, the LED of the key will function as a tri-color LED to show
status of the Station. If you assign a feature that does not require any LED activity, like Call Pick-Up, the LED of this
key will remain inactive.
Volume Keys
• "+" (plus): This is the increase key, to raise the volume of speech while talking and to decrease the Ringer
volume, when the phone is ringing.
• "-" (minus): This is the decrease key, to lower the volume of speech while talking and to decrease the
Ringer volume when the phone is ringing.
Headset Connectivity
The EON48P/48D-P/48S/48D-S provides two Headset interfaces: A 2.5mm Audio Jack and an RJ11 connector at
the bottom of the phone body.
So you can use any stereo headset of standard make with a 2.5 mm single connector or a stereo headset with an
RJ11 connector.
You can also assign Headset key function to any of the DSS keys. Refer the topic “DSS Keys Programming” for
instructions.
CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold
CA 2 CA 2 CA 2
CA 1 CA 1 CA 1
These key maps can be customized to match the exact requirement of individual users. Refer the topic “DSS Keys
Programming” for instructions on customizing the Key Maps.
Phone Menu
You can access the following PBX and phone features from the Menu of EON48P/48D-P/48S/48D-S:
Call Logs To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Contacts To add, edit, delete names and numbers of contacts in the Global Directory Part 1.
Call Forward To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-Forward-
Unconditional, and Follow Me.
Do Not Disturb To set/cancel Do Not Disturb on the phone, i.e. block incoming internal and external
calls.
Call Cost Display To view the cost of calls made from the phone.
Change User To change User Password (required for using certain features like Call Follow Me,
Password Dynamic Lock, DISA, Walk-In Class of Service, User Absent/Present, Hot Desk,
Voicemail) and for customizing Phone Settings.
Phone Settings To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).
To exit menu,
Or
• Go ON-Hook.
Operating EON48
Please refer the User Card for EON48 for instructions on operating the features of ETERNITY using EON.
EONSOFT
The EONSOFT is a PC-based Digital Key Phone. Based on a graphic user Interface (GUI), the EONSOFT offers all
the features of EON42 and EON48, making it a substitute for the Digital Key Phone. Its integration with the
ETERNITY obviates the need for a separate telephone instrument.
The EONSOFT can be installed on any personal computer with Windows or NT operating system.
Two PC-based DSS64 Consoles are available to be used with the EONSOFT. You can use either one or both
DSS64 Consoles.
DKP Port
The DKP port connects EONSOFT to the DKP port of ETERNITY's DKP card.
Handset Port:
The Handset port connects the Receiver of the phone, to be used for speech.
The EONSOFT has the provision for attaching a Handset. A handset with spring cord is supplied by Matrix and is to
be connected to the handset jack (RJ12) on the Dongle.
Headset connectivity
EONSOFT supports headset connectivity, providing a MIC and a Speaker interface. Any stereo Headset of
standard make, with dual connectors can be connected to the MIC and the Speaker on the Dongle.
COM Port
The COM port connects EONSOFT to a PC (COM Port).
After EONSOFT has been successfully installed on a PC and the DKP parameters have been configured, each
time you open EONSOFT, the display and keypad of the phone will appear on your PC screen.
Phone Display
The EONSOFT has a 2-line and 24-character display. In the ON-Hook or idle condition, the first line displays the
Station Number and the Station name. The second line displays the Day, Date and Time.
When there is an incoming call, the calling party's number is displayed on Line 2 of the LCD162.
The LCD messages for various call events (dial, transfer, forward, hold, etc.), for prompts, alerts, confirmation,
errors, text messages, are displayed.
Refer the topic “DSS Keys Programming” for instructions on assigning stations, trunks, features to keys.
162. Only if the Station, to which EONSOFT is connected, has been allowed CLIP facility in its Class of Service.
Dial Pad
The dial pad consists of 12 keys (non-programmable), which include the digit keys for 0, 1-9, and character keys for
* and #.
Function Keys
These are non-programmable keys on the keypad of EONSOFT which have fixed functions.
• Redial: This key is used for redialing the last external number.
• Func Key: This key is used for accessing the Phone menu.
• Adr: This key is used for accessing the Address Book. The EONSOFT provides the facility of an Address
Book that is integrated with the Standard Windows Address Book, for storing the numbers and addresses
of contacts. So, when a call is to be made, you can select and dial the desired number from the directory.
• Hold: This key is used for putting the caller on hold. This key is also used to make a selection in the Phone
Menu.
• OFF-Hook: This key is used for going OFF-Hook. It simulates lifting of the handset, pressing of the
speaker key to make or receive calls.
• ON-Hook: This key is used for going ON-Hook. It simulates replacing of the handset, pressing of the
speaker key to disconnect.
Navigation Keys
The following keys are used for navigating the phone menu:
• 'Func' key: This key is used for entering the Phone menu and to go back one level in the menu.
• Up and Down keys: The and keys function as the Up and Down keys to scroll the Menu and sub-
menu options. You can scroll up down the menu by clicking and scroll up the menu by clicking .
Speaker key
The 'Spk' key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable; you
can assign any other feature/function to this key.
F1 Help
F3 Spd
F4 Func
F6 Alt+Enter - Hold
F7 Xfr
F8 Spk
Esc ON-Hook
. (dot/period) Flash
(Up Arrow Key) Volume key. To increase volume of ringer and speech
(Down Arrow Key) Volume key. To decrease volume of ringer and speech
Tab (Tab
Backward shifting of selection
Backward)
Key Maps
EONSOFT can function as a Station for the Operator, Executive, and Hotel Attendant, also Guest (though unlikely
to be used by guests).
Phone Menu
The Phone menu is the same as EON42 and EON48.
• Press the 'Func' key repeatedly to go back one level in the menu, till you reach 'Menu'.
Or
If you want to use the Keyboard, press the Shortcut key for the desired function.
Tool Tips
You can assign labels and tool tips for the DSS keys, which are displayed to the user on mouse over. You can
assign the function of each key as Tool Tip, to help user in intuitive operation of EONSOFT.
Call Indication
Incoming Calls are indicated by:
In order for the EONSOFT window to pop up, you must have enabled the 'PopUp When Ring' option. When
this option is enabled and the EONSOFT window is minimized a new incoming call causes the window to pop
up to its full size notifying the user about the new call. When this option is enabled and the EONSOFT window
is maximized, a new incoming call is indicated by the flashing of the Title bar of the window.
When the 'PopUp When Ring' option is disabled and the EONSOFT window is minimized, a new incoming call
is indicated by the flashing of the EONSOFT Title at the bottom bar of the PC screen.
To answer the second incoming call, you may put the current call on hold.
Operating EONSOFT
EONSOFT can be operated using the keyboard and the mouse.
• A headset must be connected and 'Headset Connectivity' must be enabled in the 'DKP Parameters'.
Refer “DSS Keys Programming” for instructions.
• If you are using the keyboard instead of the mouse, press the appropriate Shortcut Keys listed above
and use the Number pad on the keyboard to dial digits.
Receiving calls
• When window pops up to indicate a call,
• Click 'Spk' key or 'OFF-Hook' Key.
• Talk.
• Click 'ON-Hook' key to disconnect.
What's this?
The ETERNITY provides a solid state Digital Output Port (DOP).
The DOP is located on “The Master Card” of the ETERNITY ME models and on “The CPU Card” of the ETERNITY
GE models. It is located on “The Door Phone Card” of the ETERNTIY PE.
The ETERNITY PE supports 3 DOPs while ETERNITY ME and GE support a single DOP each.
A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device, like a Door Lock
opener, a siren, or a hooter can be interfaced with the DOP via this DC contactor.
The DOP can be used for operating a variety of “Automated Control Applications”, such as a door lock release
device, a siren/hooter, a school bell, a water pump, sprinklers, and automated illuminations (office lights, porch and
terrace lights, glow signboards, street lights), and others.
How it works
• A gadget is connected to the DOP.
• If the DOP is being used for an Automated Control Application, it will turn on the gadget connected to it
according to the "Gadget Operation Mode" programmed.
There are 9 different Operation Modes. Refer the topic “Automated Control Applications” for an overview of
the operation modes in which the DOP may be used.
• The DOP can be turned ON/OFF by dialing the relevant Feature Command, which in turn switches ON/
OFF the gadget connected to it.
• The DOP can be operated also from a remote location from the Direct Inward System Access (DISA)
mode.
ETERNITY remembers the state of DOP during power failure. For instance, a water pump is being
controlled using DOP. If a power failure occurs while the pump is running, the Operator need not turn on
the water pump again on power restoration. The ETERNITY will remember the last state (in this case
pump on) and switch ON the water pump when power is restored.
Refer the chapter “Installing ETERNITY” for step-by-step instructions on connecting a gadget to the DOP of your
model of ETERNITY.
Do not connect devices that do not conform to the specifications of the DOP!
In the default Station Basic Feature Template 01 assigned to all extensions of ETERNITY, the feature 'DOP Turn
ON/Turn OFF' is enabled in the default CoS group 01. Thus all extensions of ETERNITY can use switch ON/OFF
the DOP.
If you want to restrict access to DOP operation to selected extensions, simply disable this feature in the default CoS
group, and follow these steps:
• Define a new CoS group with DOP Turn ON/Turn OFF enabled.
• Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
• Assign this new Template to the extensions to which DOP operation is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions on
programming.
If required you may also change the default Access Code '1174' assigned to the DOP, and assign a DSS key with
the function of DOP operation on the DKP extensions which are allowed access to the DOP. Refer the topic “DSS
Keys Programming” for instructions.
• Set the contact type for the DOP as appropriate: Normally Open/Normally Close. By default, the contact
type for the DOP is 'Normally Open'.
If you are using ETERNITY PE, you may configure the settings of DOP-1, DOP-2 or DOP-3, as desired.
• Exit SE mode.
How to use
The DOP will turn ON/OFF the gadget connected to it on the basis of a Timer or a Schedule or on instigation from
the sensor device connected to the Digital Input Port.
Irrespective of the "Gadget Operation Mode" programmed, the DOP (and hence the gadget connected to it) can be
turned ON/OFF by dialing a Feature Command at any time from an extension of the ETERNITY. This feature
command overrides the Operation mode defined for the gadget.
The extension from which you dial this feature command must have the feature 'DOP turn ON/turn OFF' in its Class
of Service.
OR
• DOP Number is the number of the DOP from 1 to 3 to which the gadget is connected.
• If using ETERNITY ME/GE, dial 1 for DOP Number. If using ETERNITY PE, dial the number of the
DOP which you want to turn ON/OFF.
OR
• Dial 1072-020
• Enter DOP number on the prompt on the phone's display.
• You get the confirmatory message 'DOP turned on' on your phone's display and a confirmation tone.
• Go idle.
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
OR
• Dial 1072-019
• Enter DOP number on the prompt on the phone's display.
• You get the confirmatory message 'DOP turned off' on your phone's display and a confirmation tone.
• Go idle.
What’s this?
• DDI is an ISDN Service which allows the caller to call the user on an ISDN compatible PBX or private
network directly without operator intervention.
• Using the DDI feature of ISDN, the calls can be made to land directly on the desired stations.
• The T1E1PRI and BRI trunks must be assigned a IC ReferenceNumber and OG Reference Number which
in turn defines the translation logic to handle an IC/OG. For more details refer “DDI Routing Table”, “IC
Reference Table” and “OG Reference Table” corresponding topics.
• Each ISDN Trunk is given an Installation Number by the SP. This is the combination of Main Number (MSN
No.) and the DDI Number. The Number is of max.16 digits. This is also known as ISDN Installation
Number.
• The MSN number is given by the SP whereas the Directory Numbers can be selected by the User.
However the number of digits to be used for the Directory Number should be informed to the SP.
• Please refer the topics “DDI Routing Table”, “T1E1 Trunks” and “ISDN-BRI” for details on programming.
DDI Routing is not supported on T1/E1 trunk line if you have selected E&M as the Signal Type.
How it works
Incoming Call
• When the call lands on the ISDN trunk of the PBX, the PBX checks if CLI based routing is enabled on the
trunk. If Yes, the call is routed accordingly. If CLI based routing is not enabled, then the PBX checks the IC
Reference Number assigned to the Trunk.
• If the IC Reference Number assigned to the trunk is 00, then the system further follows the logic of
Incoming Call management. For more details please refer the topic “Incoming Call Management”.
• If any other number is assigned as IC Reference Number to the trunk, the PBX searches the different IC
reference tables for a match. When a match is found, the system matches channel number of the trunk
with the channels of the table. If it does not match then the next table with the same IC reference number
is searched.
• When the channel number matches, the system uses the DDI Flexible Reference No. of the table to
identify the DDI Flexible Number table. This table helps the system to route the call to the target station.
The system first compares the received DDI number (called party number) with the DDI numbers
programmed in the DDI Flexible Number table. If a match (DDI Number) is found, the system goes ahead
with further interpreting the translation logic in the IC reference table else the system searches the next
matching DDI Flexible Number table and repeats the above procedure.
• Once the station is identified the system checks the DDI IC routing flag of the station (Please refer the topic
“Station Advanced Feature Template” for more details) on which the call is to be routed. If the flag is
enabled the call lands on the station else the call is routed to the TLG assigned to the Trunk.
• When the call lands on the DDI station, the caller gets the Ring Back Tone. The station rings for time=DDI
Timer, If the call is not answered the system checks for the Route when No reply flag in the IC reference
Table. If it is enabled the call is routed to the TLG programmed in the IC Reference Table (Trunk
Template) else the call is disconnected.
• When DDI station being called is busy, the caller gets the busy tone. The system checks for the Route
when busy flag in the IC Reference Table. If it is enabled the call is routed to the TLG programmed else
the call is disconnected.
• When the call is answered the system checks for DDI OG flag. If the flag is disabled the system does not
send the answering party number to the network. If the flag is enabled the system prepares the OG
number (Answering party number) and sends it to the ISDN Network.
• Depending on the Channel Number and the port grabbed, the OG reference number assigned to the port
is identified.
• After the OG reference number is identified, the ISDN installation no. is identified from the OG reference
table.
• The DDI Flexible reference number is also identified from the OG reference table and this helps in
identifying the flexible number of the calling station. The equivalent DDI number is found out from the
flexible number. The DDI number replaces the last digits (Number of DDI digits parameter in the DDI
Flexible Number table) of the ISDN installation number. This forms the answering/calling party number.
This is sent to the ISDN network.
• When an OG is made by a DDI station, the MSN Number + DDI number of the station is sent in the calling
party field.
• When an outgoing call is made by a Non-DDI station, the MSN Number + the first DDI number of the ISDN
=Trunk is sent in the calling party field.
Relevant Topics:
• “DDI Routing Table”
• “ISDN-BRI”
• “Trunk Feature Template”
• “T1E1 Trunks”
• “OG Reference Table”
• “IC Reference Table”
What’s this?
Direct Inward Dialing (DID) is an auto attendant feature that allows external callers to reach an extension directly
without the intervention of the Operator.
If DID is enabled on a trunk, whenever an external call lands on that trunk, the Built-In Auto Attendant or the Voice
Mail Auto Attendant of ETERNITY (if Voice Mail System card is installed) greets the caller and prompts the caller to
dial the desired extension number. The call is then placed to the extension number dialed by the caller.
ETERNITY offers Delayed DID, whereby incoming calls routed to the Operator or the Trunk Landing Group, can be
answered by the Built-In Auto Attendant or the Voice Mail Auto Attendant, if none of the landing extensions
answers the call within a certain time period.
Regular callers who know the extension numbers of ETERNITY, can use DID to reach the desired extensions
without Operator assistance. Thus, DID reduces call traffic on the Operator extension, saves callers the time for call
set-up and transfer. DID is particularly useful during non-working hours and holidays, and it helps project a
professional image of the organization.
DID will not work, when the dialed extension has Privacy from DID enabled in its Class of Service. So, if
you want to prevent external callers from accessing certain extensions, you must enable Privacy from
DID in their Class of Service. To know more, see “Privacy”.
How it works
DID can be configured on all trunk types, for the three time zones (working hours, break hours and non-working
hours).
When configuring DID on a trunk, you may choose to have calls answered by the Built-In Auto Attendant or the
Voice Mail Auto Attendant.
• The system waits for the period of the DID Answer Wait Timer (default: 05 seconds) to answer the call
during this period. The caller gets Ring Back Tone from the CO.
• The system greets the caller with the pre-recorded voice message called the DID Welcome Greeting for
the current time zone. A Voice Module must be assigned for the DID Welcome Greeting.
If no voice module is assigned as Welcome Greeting, the system will play music-on-hold after answering
the call. It will play music-on-hold until the end of the DID Music Timer (default: 5 seconds).
The DID Dial Message is played once and the caller gets Beeps. The system waits for the DID Beeps
Timer (default: 10 seconds) to expire.
• If the caller does not dial any number before the DID Beeps Timer expires, the system plays the DID Call
Transfer to Operator message and transfers the call to the Operator.
The system waits for the duration of the DID Inactivity Timer (default: 60 seconds) for the Operator to
answer the call. If there is no answer at the end of this timer, the system releases the trunk.
If the caller fails to dial digits, you can have the call disconnected instead of having it routed to the
Operator. For this, you need to enable the Disconnect DID call, when caller does not dial any digit flag
in the System Parameters. When this flag is enabled, the system will play the DID No Dial Voice message
to the caller. If the caller fails to dial a digit within the DID Beeps Timer, the system will disconnect the call.
• If the caller dials the extension number, the system checks if the number is valid.
If the dialed digits are invalid, the system plays the Wrong Dial voice message to the caller. This message
is played once. The system waits for the duration for the DID Error Tone Timer (default: 5 seconds).
If the Wrong Dial Voice Message is not programmed, the system plays Error Tone to the caller for the
duration of the DID Error Tone Timer, followed by the DID Dial Prompt.
• If the number dialed by the caller is valid, the system checks if the dialed extension is free.
• If the dialed extension is busy, the system plays the DID Busy Message to the caller. The message is
played once.
• If no DID Busy Message is programmed, the caller will hear Busy Tone. The Busy Tone is played for
duration of the DID Busy Tone Timer (default: 15 seconds), followed by the DID Dial Prompt.
To have the call disconnected if the dialed extension is busy, you may enable the Disconnect DID Call,
when dialed number is busy flag in the System Parameters.
• The dialed extension is free. The system calls the extension and plays DID Ring Back Tone Message (if
programmed) or Ring Back Tone to the caller. This message is played until the dialed extension is ringing.
• The system waits for the period of the DID Ring Timer for the dialed extension to answer the call.
• When the dialed extension answers the call, the caller gets connected to the extension.
If the dialed extension does not answer before the expiry of the DID Ring Timer, the system prompts the
caller to dial again with the DID Dial Prompt message to the caller.
• The system diverts the call to the Operator. When the call is transferred to the Operator, the system plays
the DID Call Transfer to Operator voice message (if programmed) or plays Ring Back Tone to the caller.
When the Voice Mail Auto Attendant of ETERNITY is selected as the destination for incoming calls on a trunk, this
is how DID will work:
• The Voice Mail System (VMS) installed in the ETERNITY answers the call.
• The VMS greets the caller with the Welcome message and the Greeting Message selected for the current
time zone (working hours, break hours and non-working hours).
• The VMS plays prompts to the caller to process the call further.
Delayed DID
You can use Delayed DID to have incoming calls that are not answered by the landing destinations—the Operator
and the Trunk Landing Group—within a certain time period, to be handled either by the Built-In or the Voice Mail
Auto Attendant.
• as a call lands on a trunk, the system checks the incoming call routing configured for the current time zone
for the trunk.
• on finding Delayed DID enabled, the system rings on the destination extensions (Operator and Trunk
Landing Group) for the duration of time defined for ringing the extensions (default: 10 seconds).
• if no reply is received from the extensions, the system routes the call to the auto attendant you selected,
which may by the Built-In Auto Attendant or the Voice Mail Auto Attendant.
How to configure
1. Make a list of the trunks by their port type (TWT, Mobile, SIP, T1E1PRI, BRI) and port number on which
you want to use the Built-In Auto Attendant.
2. In the “Trunk Feature Template” assigned to these trunks, enable DID for the desired time zones by
selecting DID ‘ON’.
• DID Welcome Greeting: Played to callers when answering the DID call. Different welcome greetings
can be programmed for Working Hours, Break Hours and Non-working Hours. The DID Welcome
Greeting message is played once.
• DID Dial Prompt: Played after the Welcome greeting message to prompt the caller to dial the desired
extension number. This message is played once.
• DID Ring Back Tone: Played after the caller has dialed the number and the system is ringing the
dialed extension. This message is played continuously as the dialed extension rings.
• DID Wrong Dial message: Played when the caller dials a wrong number or the number dialed by the
caller does not match with any extension number of ETERNITY. This message is played once.
• DID Destination Busy: Played when the dialed extension is busy. This message is played once.
• DID Destination No Reply: Played when the dialed extension does not respond. This message is
played once.
• DID No Dial: Played when the caller has not dialed any number. This message is played once.
• DID Call Transfer to Operator: Played to the caller when the call is being transferred to the Operator.
This message is played once.
Pre-recorded DID voice messages are provided in .WAV file format on the CD-ROM provided to you
with the ETERNITY.
The default Voice Module numbers assigned to DID messages and the messages recorded on each
module are:
Voice
Module Voice Message Application Voice Message
Number
06 DID Welcome Greeting for Night time (Non- Welcome! I am sorry, we are closed.
working hours)
08 DID - No Dial message Sorry! You have not dialed any number.
12 DID - Destination No Reply message The person you dialed is not responding.
13 DID Call Transfer to Operator message Please hold, transferring your call to the
Operator.
You may customize these DID voice messages by recording messages of your choice and assigning
them to the voice modules. For instructions on recording messages on the voice modules and
assigning voice modules to different functions, see “Voice Message Applications”.
If you do not use any of the above voice modules, the system will play the Call Progress Tone for each call
state.
1. Make a list of the trunks by their port type (TWT, Mobile, SIP, BRI, T1E1PRI) and port number on which
you want to use the Voice Mail Auto Attendant for DID.
• enable DID by selecting DID ‘ON’ for the desired time zones.
• assign the Voice Mail Group Number (default: 96) to the Trunk Landing Group.
• Configure Welcome and Greeting messages. You may either use the default, pre-recorded welcome
messages of the VMS, or record the custom welcome messages that meet your requirements, in .WAV
file format.
For more information and instructions, see the ETERNITY VMS Card System Manual.
1. Make a list of the trunks by their port type and port number on which you want to enable Delayed DID.
• if you want to use the VMS Auto Attendant for Delayed DID, assign the Voice Mail Group number
(default: 96) to the Trunk Landing Group and complete the voice mail related configuration. For
more information and instructions, see the ETERNITY VMS Card System Manual.
• if you want to use the Built-In Auto Attendant for Delayed DID, you must assign the Voice Modules
for DID Messages, as described earlier.
To know more about these timers and for configuration instructions, see “System Timers and Counts”.
You may also configure the following DID related flags, as required:
• Disconnect DID Call, when dialed number is busy: When this flag is enabled, if the dialed extension
is found busy, the system will disconnect the DID call instead of routing it to the Operator. Default:
disabled.
• Disconnect DID call, when dialed number is not responding: When this flag is enabled, if there is
no reply from the landing destination extensions, the system will disconnect the DID call instead of
routing it to the Operator. Default: disabled.
• Disconnect DID call, when caller does not dial any digit: When this flag is enabled, if the caller fails
to dial a digit within the DID Beeps Timer, the system will disconnect the DID call instead of routing it to
the Operator. Default: disabled.
These flags may be used in Hotels that provide 'Limited Services' and do not want to receive unanswered/busy
calls on the guest phones.
What’s this?
With Direct Inward System Access (DISA) remote users can access and use the system's features and facilities
using Trunks, on which this feature is enabled.
All these can be done as if being done from a local extension of the ETERNITY.
This feature allows access to system resources to remote users, and therefore has serious implications for
your system's security. Protect your system from unauthorized access and misuse.
DISA Variants
ETERNITY offers three types of DISA, each with a different method of authentication and level of access:
Callers are authenticated and allowed to use the extension on which they are logged in.
The callers must dial special digits or codes to go On-hook, Off-hook. They are allowed to make as many trunk calls
and internal calls for as long as they remain logged in to the DISA mode.
To end the DISA login session, callers must dial the Termination code or disconnect from the remote end.
Callers can access an extension to use DISA PIN Authentication-Multiple Calls only if the extension has DISA
feature enabled in its Class of Service.
Callers are not required to dial any DISA Login Code or any password.
When a caller is authenticated on the basis of CLI, the system plays the ('internal' system) Dial Tone to the caller.
To end the DISA login session, callers must dial the Termination code or disconnect from the remote end.
For this type of DISA, the DISA CLI Authentication Table must be configured first.
When the caller is authenticated on the basis of CLI, the system gives the caller direct access to the Outgoing
Trunks selected for TAC-1 for the current time zone (working hours, break hours, non-working hours)’ in the
“Station Basic Feature Template” assigned to the Auto Login extension. It plays the dial tone.
Callers are allowed to make a single external call. The system ends the DISA session on the completion of the call
by the caller or by the other remote party
For this type of DISA, the DISA CLI Authentication Table must be configured first.
To make another call, the caller must enter the DISA mode again, by calling the ETERNITY from the remote
location.
DISA with CLI Authentication - One Call Answer Signaling is generally used when ETERNITY is deployed in the
Gateway Mode, where ETERNITY is configured to send an answer signal to the caller/calling device, receive the
DTMF digits dialed by the caller/calling device and dial out the digits dialed by the caller/calling device. To know
more, see “Gateway Application-Answer Signaling”.
How it works
For this feature to work, you must enable the desired DISA variant on the desired trunks: TWT, Mobile, SIP,
T1E1PRI, BRI.
• The system checks if a DISA variant is enabled on the trunk for the current time zone, i.e. working hours,
break-hours and non-working hours.
• If a DISA variant is enabled on the trunk, the system processes the call according to the DISA variant
enabled on the trunk.
• The caller must dial the DISA Login Code consisting of:
• the DISA Feature Access Code.
• the number of the extension the caller wants to access.
163. If no voice message is recorded, the system plays music-on-hold to the caller.
• On successful login, the system starts the DISA Idle State Timer (configurable; default: 20 seconds).
The system waits for the caller to go Off-hook164.
• When the caller goes Off-hook by dialing the Off-hook code #1, the system plays the internal dial tone
and waits for the caller to dial digits.
• If the caller dials an external number using a TWT trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes)165.
• The system waits for the caller to dial digits within the DISA Inactivity Timer.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
• The caller can make as many trunk calls and internal calls.
• The caller can terminate the DISA login session either by disconnecting from the remote end or by
dialing the Termination Code #9.
• The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
• If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table166.
• The caller gets logged into the Auto Login extension and gets the dial tone of ETERNITY.
• At the end of the call, the caller dials the On-hook code #0 to go On-hook. To make another call, the
caller dials Off-hook code #1 and dials the desired number. Thus the caller dials the On-hook and Off-
hook codes to make as many trunk and internal calls as desired.
• If the caller dials an external number using a TWT trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes)167. The system waits for the caller to dial digits within the DISA
Inactivity Timer.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
164. If the caller does not go Off-hook within this timer, the system releases the call.
165. DISA Inactivity Timer is not applicable for T1E1PRI lines, BRI lines, SIP and Mobile trunks.
166. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Routing configured in
ETERNITY.
167. DISA Inactivity Timer is not applicable for T1E1PRI line, BRI, SIP and Mobile trunks.
• The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
• If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table168.
• The caller gets logged into the Auto Login extension and gets dial tone of the outgoing trunks selected
for TAC-1 for the current Time Zone (working hours, break hours, non-working hours).
• If the caller dials an external number using a TWT trunk, the system starts the DISA Inactivity Timer
(configured; default: 2 minutes).
• The system waits for the caller to dial digits within the DISA Inactivity Timer. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
• After the external call is completed, i.e. the caller disconnects from the remote end or the other remote
called party has disconnected, the caller is logged out.
• To make another external call, the caller must call the DISA enabled trunk of ETERNITY again.
In all the variants of DISA, the caller can use all the features allowed in the “Class of Service (COS)” of the
extension the caller is logged in to (using PIN Authentication or CLI Authentication).
• DISA calls in the SMDR report are marked as "O" in the remarks column. See “Station Message Detail
Recording–Report”.
• If DISA is disabled, ETERNITY will route the call by DID logic, if DID is enabled. If DISA and DID both
are disabled, the incoming call will be routed as per the incoming call routing configured. To know
more, see “Direct Inward Dialing (DID)”.
How to configure
To provide DISA to remote users you need to do the following configuration:
• Select the DISA variant for the Trunks on which you want to apply this feature in their “Trunk Feature
Template”.
• Enable DISA in the “Class of Service (COS)” of the extensions which you want to allow callers to access
using DISA.
• Change the User Password of the DISA extensions, if you selected DISA PIN Authentication-Multiple
Calls. If you selected DISA PIN Authentication-Multiple Calls on a trunk, the default User Password (1111)
will not work. See “User Password” and “System Security” more information and instructions.
168. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Routing configured in the
ETERNITY.
• If you have selected the DISA CLI Authentication-Multiple Calls or CLI Authentication-One Call Answer
Signaling on a trunk, you must configure the CLI Authentication Table.
• Make a list of remote users and their numbers whom you want to allow DISA.
• For each remote user’s number on your list, write the Extension number of the ETERNITY you want to
allow this extension user to log in.
• Open Jeeves.
• You can configure as many as 999 numbers in this table, by clicking the tabs of the index on the top of the
table.
• Refer to the list of remote user numbers and the corresponding ETERNITY extension numbers you made.
• In the Calling Number column, enter the number of the remote users whom you want to allow access to
DISA using CLI Authentication. The system will match the CLI of the callers with the numbers you store
here.
• For each Calling Party Number, in the Auto Login as field, select the extension Port Type (SLT, DKP, SIP
Extension, ISDN Terminal) and Port Number you want to allow access to after the Calling Party Number
is authenticated.
• 4111-Index-Calling Number-#*
Where,
Index is from 001 to 999.
Calling Number may contain a maximum of 16 digits.The allowed digits are 0-9, #, *, A, B, C, D, +. Use
following codes to enter these digits:
A #4
B #5
C #6
D #7
+ #8
* **
# ##
Port
Port Type Meaning
Number
00 000 None
28 01 to 64 ISDN Terminal
For example, to configure extension '3001', which is a DKP with port number 001, as auto login station in
Index 001 of the Table, you must dial 4112-001-02-001.
• Exit SE mode.
• the number of the Trunk on which DISA is enabled and the variant of DISA enabled on this trunk.
• the number of the extension and the user password which you want to access, if using DISA with PIN
Authentication.
• the duration of the DISA related Timers: The DISA Idle State Timer and the DISA Inactivity Timer, so that
you may dial digits accordingly, without delay.
However, ETERNITY will not be able to understand the conventional way of dialing 'flash' key or going on-hook with
momentary make/break of loop current. Therefore, ETERNITY supports specific codes for specific activities. If
these codes are received during a DISA session, ETERNITY interprets it and performs the associated activity.
When you are in DISA mode, use the following codes to indicate an activity:
on-hook #0
off-hook #1
Flash #2
Pause #3
A #4
B #5
C #6
D #7
+ #8
# ##
End of String #*
To use DISA,
• Dial the number of the Trunk on which DISA is enabled for the current time zone, Working, Break, Non-
working hours.
• ETERNITY answers the call. You will get music or DID Voice Message, if configured.
The features listed below are not supported in the DISA mode.
• Auto Call Back
• Auto Redial
• Call Park
• Call Chaining
• Self Ring Test
• Trunk Reservation
• Walk-In Class of Service
• Live Call Supervision
The Direct Station Selection (DSS) Console is a two-wire digital terminal with 64 to 72 keys (depending on the
model) with single and dual-/tri-color LEDs. The DSS Console is an add-on module for the Digital Key Phone
(DKP), functioning as the extension of the DKP. It provides you quick access to Stations, Trunks, Features/
Functions of the ETERNITY or at the touch of a single key, making call operations easy.
While the DSS Console is more commonly used by the Operator/receptionist in an organization, it meant for use by
anyone who needs to access the many features of the ETERNITY at a single touch of a button.
The Matrix DSS Consoles are available as DSS64 (64 keys) and DSS72 (72 keys), and have different appearance
to match the style of each model of the digital key phone, EON.
Two DSS Consoles can be attached to a single DKP. Each DSS Console occupies a Digital Key Phone Port. For
example, if you attach two DSS consoles to a single DKP, three DKP ports would be occupied.
The DSS Console can be attached with the ETERNITY in the same way as the DKP, EON, and is programmed as
an attachment of the DKP (Refer the sections “Installing DSS Consoles” and “Programming DSS Console Keys” for
instructions).
You can attach two DSS consoles to a single DKP. This may be necessary, if you want to access most or all of the
features/functions of the ETERNITY at a single touch of a key.
When a single DSS64 is attached with a DKP, the DSS keys of the DKP as well as all the 64 keys of the DSS64 can
be used. If the DSS72 is used, 72 keys can be used as DSS key, in addition to the DSS keys on the DKP. Similarly,
if two DSS64 are attached to a DKP, 128 additional keys are at your disposal to be used as DSS keys.
Each DSS Console that is attached to a DKP occupies a DKP port. Hence, the more DSS Consoles you attach to
DKPs, the lesser number of DKP ports will be available on the ETERNITY.
01 17 33 49
01 25 49
02 18 34 50 02 26 50
03 19 35 51 03 27 51
04 20 36 52 04 28 52
05 21 37 53 05 29 53
06 22 38 54 06 30 54
07 23 39 55 07 31 55
08 24 40 56 08 32 56
09 25 41 57
09 33 57
10 26 42 58
10 34 58
11 27 43 59
11 35 59
12 28 44 60
12 36 60
13 29 45 61
13 37 61
14 30 46 62
14 38 62
15 31 47 63
15 39 63
16 32 48 64
16 40 64
17 41 65
18 42 66
19 43 67
20 44 68
21 45 69
22 46 70
23 47 71
24 48 72
You can assign Station numbers or features/functions to the keys on the DSS Console in the same way as you
would assign functions to the DSS keys of various models of EON, so that they can be accessed easily simply by
pressing a single key.
LEDs
Each DSS Console key is equipped with an LED which glows in single (Red) or in tri-color (Green, Red, Orange)
depending on the function assigned to it.
When a Station or Trunk is assigned to a DSS Console key, the LED functions as a tri-color LED to show the status
of the Station (whether ringing, busy, in speech, on hold). When a Feature is assigned a DSS Console key, it
functions as a single color LED to indicate whether the Feature has been accessed or activated (for example:
whether the feature is set or canceled).
The LED color and cadence of the DSS Console keys is the same as that of the DSS keys of EON42, and 48.
Remember, as not all Features/Functions require an LED, the LED of the DSS keys function only if the LED is
relevant for the feature/function which is assigned to the keys.
What's this?
Distinctive Rings are ringing patterns used for distinguishing between different types of call events.
1. Internal Call
2. Priority Internal Call
3. External Call
4. Alarm Call
5. Auto Call Back Call
6. Auto Redial Call
7. Message Wait Call
8. SE Mode (Programming Ring)
9. Operator Alarm
10. Emergency
11. Self Ring
12. Call Supervision
13. Door Phone Call
14. Presence
With Distinctive Rings, it is possible to use ring cadence of user's choice for each of these call events. For instance,
Triple ring can be set for 'Priority Internal Calls' and long rings can be set for 'Alarm Calls'.
A set of ring types is called Distinctive Ring type. The default Distinctive Ring Types are:
Call Event Ring Type set 1 (T1) Ring Type set 2 (T2) Ring Type set 3 (T3)
Auto Redial Very Long Slow Very Long Slow Very Long Slow
Message Wait
Short Fast Short Fast Short Fast
Call
Programming
Continuous Continuous Continuous
Ring
Double 400-200-400-2000
400-200-400-200-400-
Triple
2000
How it works
At the time of installation, when the System Engineer selects the “Region” (as per the geographical location of the
system), ETERNITY loads the country-specific Distinctive Ring Type defined for the selected Region.
Refer the topic “Default Settings” for the default Distinctive Ring Type applied to your country/region.
Demonstration of rings
It is possible to demonstrate Ring Types to users by dialing the SE commands from EON or an SLT.
By default, the system will play each Ring Type as demonstration for 30 seconds.
How to configure
The country-specific Distinctive Ring Pattern is set automatically by the system when you select the Region Code
and issue command to default the system. However, if required, the System Engineer can change the default Ring
Pattern loaded by the system.
• Select the desired Ring Pattern for each call event that you want to customize.
• Exit SE mode.
How to use
Users of ETERNITY may be acquainted with the different Distinctive Rings played by the system so that they can
associate the terms used to describe the rings with the sound emitted by the system for each ring.
Ring Patterns can be demonstrated to users by dialing the SE commands from EON or an SLT.
01 for Continuous
02 for Short Fast
03 for Short Long
04 for Short Very Slow
05 for Long Fast
06 for Long Slow
07 for Very Long Slow
08 for Double
09 for Triple
• You get the prompt 'Go Idle for Ring' on your phone display.
• Go Idle.
01 for Continuous
02 for Short Fast
03 for Short Long
04 for Short Very Slow
05 for Long Fast
06 for Long Slow
07 for Very Long Slow
08 for Double
09 for Triple
• Exit SE mode.
What's this?
Extension users may wish to restrict calls to their extensions in order to work uninterrupted by frequent phone calls.
The feature, Do Not Disturb, enables users accomplish this.
This feature is useful to extension users who are in the middle of a meeting or any important work that requires their
undivided attention. This feature can be used in hotels and hospitals, this feature can be used to restrict calls made
to guests and patients. See ETERNITY Hospitality System Manual.
• Extension users
• Operator for extension users, referred to as DND-Remote.
Doing so, calls from other extensions will be barred. However, the extension user would continue to receive:
• all the external calls transferred by the Operator/any other phone, DID Calls and DDI Calls.
• Alarm calls.
• Reminder calls.
• Auto Call Back calls.
DND has three supplementary features: “DND-Override”, “DND Text Message” and “Voice Message for DND
Notification”.
DND-Override
As the feature title suggests, 'DND-Override' allows the caller to land on the called extension, despite DND set on
the extension.
DND-Override will not work if the called extension has 'Privacy from DND-Override' enabled in its Class of Service.
When setting DND (also DND-Remote), the extension user/Operator can select an appropriate text message to be
displayed to the calling extension.
This DND text message is displayed on the calling extension, but only if the calling extension is the proprietary
digital key phone, EON.
Voice Message Notification for DND is particularly useful when the phone on which DND is set is an SLT.
When DND is set on an extension of ETERNITY, callers who try to reach that extension are played an error tone.
Callers who are using EON are displayed the DND Text Message set by the called extension, and thus come to
know the cause of the error tone. Such a facility is not available to callers who are using SLTs, who can hear only
the error tone and have no way of knowing the cause of the error tone.
Hence the feature Voice Message for DND Notification, whereby a pre-recorded Voice Message notifies them of
the DND set on the called extension.
To play to the callers pre-recorded voice messages as DND Notification, you must record and assign a Voice
Module.
How it works
A, B and C are extension users.
B has EON, while C has an SLT.
B has DND-Override in his Class of Service, C does not have this feature.
DND Text messages as well as Voice Message Notification for DND have been programmed by the System
Engineer.
• A has set DND on his extension with the DND Text message 'In Meeting'169.
• B calls A.
• As B has DND-Override, the Voice Message for DND Notification is played to B once, and the DND
message 'In Meeting' set by A appears on B's phone display. B gets routing Beeps.
• To exercise DND-Override, B must dial '4' the feature access code for 'DND-Override' during either during
the Voice Message or during the routing Beeps.
• B gets Ring Back Tone, if A's extension is free.
• B gets Busy Tone, if A's extension is busy.
• However, if A has Privacy from DND Override, B will get error tone and the DND message set by A
appears on B's phone.
If B fails to dial the DND-Override code before the end of the routing beeps, error tone will be played to
him.
• C calls A.
• As C has an SLT, C will get only the Error tone.
• But as Voice Message for DND Notification is programmed in the system, C will be played the pre-
recorded message once.
169. While DND and DND Text Message can be set from any phone, DND Text Message can be viewed on EON only.
How to configure
For this feature to work, 'DND', 'DND-Override' and 'Privacy from DND-Override' must be enabled in the Class of
Service group of the extensions which is to be allowed this feature.
Besides these, the System Engineer may program the DND Text Message and the Voice Message for DND
Notification, as per user requirements.
While it makes sense to offer all stations DND, providing DND-Override and Privacy from DND also to all stations
will not serve the purpose of DND.
Decide which stations are to be allowed 'DND', which are to be allowed 'DND-Override', and which are to be
allowed 'Privacy from DND-Override'. Generally, DND-Override is allowed to the Operator station. It may be
allowed to stations of persons in senior positions in the organization. Similarly, Privacy from DND-Override may be
allowed to persons in senior positions in the organization.
If you want to allow DND to all stations, retain the default CoS group 01 in Station Basic Feature Template 01.
However, if you want to allow DND only to selected stations, disable this feature in the default CoS group 1.
Similarly, if 'DND-Override' is to be to be allowed to the Operator and a few other stations, follow these steps:
1. Define a CoS group with DND-Override enabled. If DND is also to be allowed, enable both DND-Override
and DND in this CoS group.
2. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
3. Assign this newly prepared Station Basic Feature Template to the Operator station on which 'DND-
Override' is to be enabled.
Repeat the same steps to allow 'Privacy from DND-Override' to selected stations. For stations that are to be
allowed 'DND' as well as 'Privacy from DND-Override', enable both features in the CoS group in the Station Basic
Feature Template applied on these stations.
Similarly, for stations that are to be allowed 'DND', 'DND-Override' and 'Privacy from DND-Override', enable all
three features in the CoS group that you prepare for these stations.
1 Do Not Disturb
2 Unavailable
3 In Meeting
4 In Conference
5 Try on Mobile
6 On Vacation
On Business
7
Trip
8 Out of Office
9 With a Guest
You can use these default message options or program messages as per user preferences.
• All the default text messages appear in the DND message field. Change the DND text messages as
required. Click the field and enter your custom DND text message.
• To reload default DND text messages, dial command 1501. The default DND Text Messages are given
above in the table.
• Exit SE Mode.
• Also, refer the topic “Digital Key Phone-Operation” for instructions on entering alphanumeric characters
using the keypad of EON.
Record a Voice Module with the message "The dialed extension has activated Do Not Disturb" (recommended).
Assign the Voice Module to the Voice Message Application number 44 defined for 'DND Notification'.
How to use
OR
• Dial 18
• Scroll to select from any of the DND messages that appears on the phone's display:
• Do Not Disturb
• Unavailable
• In Meeting
• In Conference
• Try on Mobile
• On Vacation
• On Business Trip
• Out of Office
• With a Guest
• Dial 18-0
• You get a text message 'DND Cancelled' on the phone's display and confirmation tone.
To cancel DND:
• Lift handset.
• Dial 18-0
• Replace handset.
DND-Remote
• Do Not Disturb
• Unavailable
• In Meeting
• In Conference
• Try on Mobile
• On Vacation
• On Business Trip
• Out of Office
• With a Guest
To cancel DND-Remote,
DND-Override
What's this?
A Door Phone is typically used for monitoring an entrance door. It is installed in place of the Doorbell.
The door phone is similar to any ordinary phone; except it does not have a hook-switch or a dial pad. Usually, it is a
weather tight box, equipped with a button like a doorbell, which visitors press.
ETERNITY offers the Door Phone feature exclusively on its ETERNITY PE model and its variants.
When visitors press the Door Phone Call Button, the phone programmed to receive the call (landing destination)
rings. The user of the called phone can answer the door phone call by simply lifting the handset and talk to the
visitor at the door. The user of the called phone can have the door opened for the visitor by either physically
appearing at the door or operating a door lock release device.
The Door Phone feature of the ETERNITY PE allows users to operate the door phone from a remote location (off-
premises) by having their calls routed to an external number.
The Door Phone feature of the ETERNITY PE is very convenient to have at:
• Delivery entrances: It is not necessary to have company personnel monitor delivery entrances. They can
just answer the Door Phone instead.
It is also possible for the doctor/pharmacist to have calls of the door phone landed on their mobile number
or any other fixed line external number, and open their clinic/pharmacy from their current (remote) location
to let the patient in.
• Residences and Apartment entrances: The identity of the visitors can be screened before letting them
in. The occupants of the house can greet their guests/relatives and let them enter the house even in their
absence by answering the door phone from their current (remote) location and opening the door for them.
Matrix does not supply Door Phones, but only the Interface to connect Door Phones. Any standard 4-wire
Door Phone can be connected to the ETERNITY PE.
How it works
The Pre-requisites
• A four-wire Door Phone connected to the Door Phone Port on the Door Phone Card of the ETERNITY PE.
ETERNITY PE supports 3 Door Phone Ports, so you can connect as many four-wire Door Phones. This
may be required in buildings with more than one entrance.
ETERNITY PE supports 3 Digital Output Ports, allowing you to connect as many Door Lock devices.
The Process
• When a visitor arrives at the entrance and presses the door phone button, the system senses the doorbell
and places the call on the phone programmed as the landing destination. The destination phone may be
extensions of the ETERNITY or an external fixed line or mobile number.
• The ETERNITY plays a distinct Ring Type on the extension, to indicate to the extension users that it is a
door phone call. The Ring Type is programmable; by default Triple Ring is set as ring type.
• The visitor is played Ring Back Tone while the destination extension rings for the duration of the 'Door
Phone Ring Timer (programmable)170.
If the destination extension does not answer the call within this Timer, the Door Phone call is dropped and
the Door Phone goes idle.
• When the destination extension answers the call, the Door Phone circuit is activated and two-way speech
is established between the visitor and the extension user.
• The extension user may now either physically appear at the door to open it, or dial the Open Door Lock
Code (default: 1173) from the current extension.
• the extension user puts the call on hold, music-on-hold is played to the visitor.
• the extension user dials the Feature Access Code to 'Open the Door' (default: 1173).
• the Door Lock is opened for the duration of the Timer 'Open Door for Time, (programmable; default: 5
seconds)171,172.
• as soon as this Access Code is dialed, speech is reestablished with the visitor.
• the extension user invites the visitor to enter the building.
• The Door Lock closes on the expiry of the "Open Door for" timer.
• The Door Phone goes idle only when the extension user goes ON-Hook.
• The extension user can open the Door Lock also by dialing the feature command to operate the DOP.
For this, however, the extension must have the facility "DOP Turn ON/Turn OFF' in its “Class of Service
(COS)”. Refer the topic “Digital Output Port (DOP)” for operation instructions.
• It is possible for the any extension of the ETERNITY PE to establish speech with the Door Phone even
when it is idle, by dialing the unique access code assigned to the Door Phone.
170. The "Door Phone Ring Timer" determines the time for which the landing destination shall ring for the door phone call. This Timer is
necessary because often visitors may press the door phone switch as they would do a door bell, for one or two seconds only,
whereas the call must remain present for a longer period of time for it to be answered.
171. This is the time for which the door will remain open.
172. An error tone will be played to the extension user for the duration of the error tone timer, if a DOP has not been assigned to the
Door phone port.
• When the visitor presses the Door Phone button, the ETERNITY makes a call to the pre-programmed
external number, which may be a fixed line or a mobile number.
The ETERNITY connects the Door Phone to the Trunk Port on the basis of the Outgoing Trunk Bundle
Group assigned to the Door Phone Port. When connected to the Trunk Port, the Door Phone receives Call
Progress Tones of the CO Network.
• When the external number answers the call, speech is established between the visitor and the external
number.
• The called party on the external number ascertains the identity of the visitor.
• the called party on the external number puts the visitor on hold by dialing '#2'. The system plays music-
on-hold to the visitor.
• the called party on the external number dials the Feature Access Code for 'Open the Door' (default:
1173)173.
• the Door Lock is opened for the duration of the Timer 'Open Door for Time, (programmable; default: 5
seconds)174,175.
• as soon as this Access Code is dialed, speech is reestablished with the visitor.
• the visitor is invited to enter the building.
• the called party on the external number disconnects the call176; the Door Phone goes idle.
This way, it becomes possible to answer the door bell call from a remote location and open the door lock
from the remote location.
• The Door Lock can be opened by the called party on the external number also by dialing the command
for operating the DOP to which the Door Lock is connected, instead of dialing the 'Open the Door'
access code. Refer the topic “Digital Output Port (DOP)”for operation instructions.
• The called party on the external number can also call the Door Phone using Direct Inward System
Access (DISA).
• It is also possible for the called party on the external number to make multiple calls, while putting the
visitor at the Door Phone on hold. For this, the party must be logged in “Direct Inward System Access
(DISA)” mode.
• The Door Phone feature of the ETERNITY PE offers the flexibility of selecting the Routing Mode for Door
Phone calls. The system supports two Door Phone Call Routing Modes:
• Manual: Whenever you want to route the Door Phone Calls, you can alternately select the landing
destination, i.e., select a routing group at one time, the external number (the programmed fixed line or
mobile number) the next time, as required. The extension user can change the Call Routing Mode viz.
routing group or external number, as and when required.
173. This access code is to be dialed only when in speech with the visitor. If this access code is dialed when there is no speech, an error
tone will be played to the extension user for the duration of the Error Tone Timer.
174. This is the time for which the door will remain open.
175. An error tone will be played to the extension user for the duration of the error tone timer, if a DOP has not been assigned to the
Door phone port.
176. If the trunk used for placing this call is a Two-Wire Trunk, dial #0 to disconnect the call.
For this, a “Time Tables” must be assigned to the Door Phone Port, defining the Time Zones, i.e.
working hours, break hours and non-working hours. Routing Group must be defined for each Time
Zone. Similarly, if external number is selected as landing destination for any Time Zone, the number
must be programmed. The ETERNITY will follow the Time Table programmed and route the calls
according to the current Time Zone to the destination phone (routing group/external number)
programmed for the current Time Zone.
• The landing destination - Routing Group and the External Number - can be programmed only by the
System Engineer (SE).
• The Routing Mode for Door Phone Calls, 'Scheduled' or 'Manual' mode can be set by the SE as well as
the extension users (User Mode). However, extension users must have the feature "Door Phone
Settings" enabled in the '“Class of Service (COS)” group assigned to their extension. With this feature
included in their Class of Service, the extension users can switch between scheduled and manual
modes by dialing the Feature Command.
• The Time Table, the Routing Group, and the External Number for routing the Door Phone Calls can be
programmed by the System Engineer only.
• The Outgoing Trunk Bundle Group assigned to the Door Phone Port used for routing door phone calls
to an external number will be common for both Scheduled and Manual modes.
ETERNITY PE supports 3 Door Phone Ports. You can connect any 4-wire Door Phones of standard make to these
ports.
How to configure
For the Door Phone feature to work, you must program the related set of Door Phone Parameters, allow 'Door
Phone Settings' in the Class of Service of extensions defined as the landing destination for Door Phone calls. If a
Door Lock is installed in conjunction with the Door Phone, you must also enable access to operate the Digital
Output Port in the Class of Service of extensions that are to be allowed to open the Door Lock.
• The parameters for Manual Door Phone Routing will appear on your screen.
• Access Codes: Each Door Phone port can be given a unique access code. The Access Code allows the
extension user to call the related Door Phone. When an extension user dials the Door Phone Access
Code, the extension user will get connected to the related door phone and can speak to the visitor.
The default access codes for Port 1, Port 2, Port 3 are 2601, 2602 and 2603. You may change these
Access Codes. The Access Code assigned to the Door Phone port may be a maximum of 6 digits.
Refer the topics “Access Codes” and “Conflict Dialing” to know more.
• Name: It is also possible to assign a 'Name' to each door phone port for easy identification. This is useful
where there are multiple entrances each having a door phone installed. The Name you assign to a Door
Phone may be a maximum of 18 characters. By default Door Phone Port1, 2 and 3 are named 'Door
Phone 1', Door Phone 2' and Door Phone 3' respectively.
• Route Door Phone Calls to: Select the landing destination for Door Phone calls: Routing Group or
External Number.
• Routing Groups: program this parameter if you have selected Routing Group as the landing destination.
By default Routing Group 01 is assigned.
• External Number: Program this parameter if you have selected External Number as the landing
destination. Enter the fixed line or mobile number to which door phone calls should be routed. The number
must not exceed 16 digits.
• OGTB Group: Define the Outgoing Trunk Bundle Group (OGTBG) through which the door phone calls to
the external number should be routed. By default OGTBG 01 is selected.
• The Door Phone parameters page will open again, with additional parameters.
• Time Table: Each door phone port can be assigned a Time Table, with defined working hours, break
hours and non-working hours, so that calls landing on Door Phone can be routed to the destination phone
(routing group or external number) according to the current time zone. By default Time Table 1 is assigned
to all Door Phone ports. You may retain the default Time Table or assign a different Time Table.
• If you want to retain the default Time Table, check if the time zones defined in the default Time Table 1
fulfill your requirement by opening the 'Time Table' link in this column.
• The Time Table page will open, make the necessary changes, if required and click 'Submit' at the
bottom of the page to save your settings.
• Return to 'Door Phone Parameters' page.
• If you want to define a new Time Table for the door phone port, open the 'Time Table' page and define
the time zones in a new Time Table, for example, 2. Click 'Submit' at the bottom of the page.
• Return to the 'Door Phone Parameters' page.
• Door Phone Call Routing Mode: Select 'Scheduled' as the mode of call routing for the door phone port.
When you select one of these options, only those parameter fields related to the option will be editable on
the page.
• Route Door Phone Calls to: define the landing destination for the door phone calls. You can select a
different landing destination - a Routing Group or an External Number - for each Time Zone, i.e. Working
Hours (WH), Break Hours (BH) and Non-Working Hours (NH).
You cannot program a different External Number for each Time Zone.
• Routing Group: program this parameter if you have selected Routing Group as the landing destination for
any of the Time Zones. By default Routing Group 01 is assigned.
• If you want to assign a different Routing Group, click the 'Routing Group' link to open the page.
• Create another routing group, for example, 04, and click 'Submit' at the bottom of the page to save
changes.
• Return the Door Phone Parameters page.
• Assign the number of the Routing Group you created (04) to the related Time Zone.
• OGTB Group: Define the Outgoing Trunk Bundle Group (OGTB) through which the door phone calls to
the external number should be routed. By default OGTBG 01 is selected.
• Door Opener: Select the number of the Digital Output Port (DOP) to which the Door Lock for the Door
Phone is connected. For example, if the Door Lock for Door Phone 1 is connected at DOP1, select this
port as Door Opener. By default, no Door Opener has been assigned.
• Open Door For: This is the Time for which the Door Lock should remain open. Program this parameter if
using a Door Lock with the Door Phone. The range of this timer is from 01 to 99 seconds. By default the
Open Door Timer is set to 05 seconds.
• Door Phone Ring Timer: This is the time for which the Door Phone will ring on the landing extension in
the Routing Group. The range of this timer is from 001-255 seconds. The duration of this Ring Timer is set
to 30 seconds by default. You may program the desired duration.
1 for Scheduled
2 for Manual
Default: Scheduled
To program Scheduled - Door Phone Call Routing for each Time Zone, dial:
• 3223-1-Door Phone-Time Zone-Call Routing Type to program time zone wise call routing for a single
door phone port.
• 3223-2-Door Phone-Door Phone-Time Zone-Call Routing Type to program the same routing type
and time zone for a range of door phone ports.
• 3223-*-Time Zone-Call Routing Type to program the same routing type and time zone for all door
phone ports.
Where,
Door Phone is the software port of the Door Phone from 1 to 3.
Time Zone is
To program the OG Trunk Bundle Group (OGTBG) for Door Phone Port to route calls on External Number,
dial:
• 3229-1-Door Phone-OGTB Group to program OGTBG for a single door phone port.
• 3229-2-Door Phone-Door Phone-OGTB Group to program the same OGTBG for a range of door
phone ports.
• 3229-*-OGTB Group to program the same OGTBG for all door phone ports.
Where,
Door Phone is the software port of the Door Phone from 1 to 3
OGTBG is from 01 to 32.
Default: 01
Default: None
• Exit SE mode.
• The option of selecting the Routing Mode for Door Phone calls, i.e. 'Scheduled' or 'Manual'.
• The facility to select the Call Routing Destination in the 'Manual Mode'.
In the default Station Basic Feature Template 01 assigned to all extensions of the ETERNITY PE, the default Class
of Service group 01 has the feature "Door Phone Settings" enabled. So, all extensions of ETERNITY PE are by
default allowed this feature.
There is no need to program this feature, if the default COS group 01 is assigned to the landing destination
extensions.
If a different COS group is assigned to the landing destination extensions, check if this feature is enabled in the
assigned COS group and enable this feature if not already included.
If you want to allow this COS feature exclusively to the landing destination extensions and deny this feature to all
other extensions, follow these steps:
Similarly, for extensions that are to be denied the 'Door Phone Settings' feature, follow the same steps, but
disable the 'Door Phone Settings' in the COS group and apply the Station Basic Feature Template on the
extensions which are to be denied this feature.
Refer the topic “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
To be able to open/close the Door Lock operating the Digital Output Port (DOP), the feature 'DOP Turn ON/OFF'
must be enabled in the Class of Service group allowed to the extension operating the DOP.
In the default Station Basic Feature Template 01 assigned to all extensions of the ETERNITY PE, the default Class
of Service group 01 has the feature "DOP Turn ON/OFF" enabled.
There is no need to program this feature, if the default COS group 01 is assigned to the landing destination
extensions.
If a different COS group is assigned to the landing destination extensions, check if this feature is enabled in the
assigned COS group. Enable this feature if not already included.
However, if you want to allow this COS feature exclusively to the landing destination extensions and deny this
feature to all other extensions, follow these steps:
For extensions that are to be denied the 'DOP Turn ON/OFF' feature, follow the same steps, but disable the
'Door Phone Settings' in the COS group and apply the Station Basic Feature Template on the extensions
which are to be denied this feature.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
How to use
To select a Call Routing Mode177:
177. Manually by the user of the extension programmed as the landing destination for the Door Phone calls.
What's this?
Dynamic Lock allows extension users to change the Toll Control Levels (Calling Permissions) of their extensions on
their own by dialing a code.
The System Administrator/Operator can also change the Toll Control Levels of extensions using Dynamic Lock.
With this feature, extension users can prevent misuse of outgoing call facility from their extensions, especially in
their absence.
Dynamic Lock also forms the basis of 'Call Privilege', which is feature of the Hotel Application of ETERNITY. Refer
the ETERNITY Hospitality System Manual to know more.
There are four types of Toll Control Levels, starting from Level 0 to Level 3 that can be set for extension phones.
For each Toll Control Level from 0 to 3, a 'Call Privilege’178 is to be assigned and corresponding numbers strings to
be allowed and number strings to be denied for each Call Privilege are to be programmed.
• Toll Control - Level 0 is Time Zone based, wherein the Call Privilege Type must be defined for each Time
Zone, i.e. Working Hours, Break Hours and Non-Working Hours. For instance, you may define
'International Calls' as Call Privilege for Working Hours, 'Local Calls' as Call Privilege for Break Hours and
'No Calls' as Call Privilege for 'Non-Working' Hours.
By default, Call Privilege 'International Calls' is selected for all three Time Zones.
• Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
• Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
• Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
The Call Privilege for each of the above Toll Control Levels can be redefined according to user
requirements. For example, Toll Control Level 3 can be programmed for allowing all types of calls by
selecting 'International Calls' as Call Privilege Type, and programming the numbers to be allowed and
denied in the 'International Numbers' List. Level 0 can be programmed to allow only Local Calls, by
programming the strings of 'Local Numbers'.
Extension users who are allowed the Dynamic Lock feature in their Class of Service, can set the Toll
Control Level in two ways:
• Manually: the extension user changes the Toll Control Level of the extension whenever s/he wants by
dialing the feature access code.
178. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, International Calls and Limited Calls.
Thus the extension user sets Dynamic Lock s/he manually selects the desired Toll Control Level for
his/her extension and restores the original Toll Control Level assigned to the extension.
• Automatically: the extension user changes the Toll Control Level of the extension using the Dynamic
Lock Timer. The user sets the Timer to the desired number of minutes. On the expiry of this Timer, the
system restores the original Toll Control Level assigned to the extension.
For example, an organization has defined Toll Control Level 0 as Local Calls, and Level 3 as
International Calls. An extension user of this organization is assigned Level 0. When this extension
user wants to make international calls, he sets the Dynamic Lock Timer and selects Toll Control Level
3. At the end of the timer, Level 3 gets locked and Toll Control Level 0 is reapplied on the extension
phone.
• The changing of Toll Control level requires the user to dial the 4-digit User Password. The system will
not accept the default User Password (1111). The extension user must first change the default User
Password.
• The Dynamic Lock Timer must be set to '00' when using Manual Dynamic Lock.
How it works
The Pre-requisites
• The Toll Control Levels 0 to 3 are programmed in the Station Basic Feature Template applied on the
extension.
The Process
OR
• The Operator sets Dynamic Lock manually for an extension by entering the extension number and
selecting the Toll Control Level.
• The Operator sets Dynamic Lock for an extension by entering the extension number, selecting the Toll
Control Level, and setting the Dynamic Lock Timer.
• Now, whenever a call is made from extension A, the system checks for Toll Control Level.
• The system then checks the associated Lists of allowed and denied numbers.
• If the Toll Control Level is 0, then Toll control is time zone based, i.e. working hours, break hours and
non-working hours. The outgoing call is allowed/denied as per the Call Privilege and the corresponding
Allowed and Denied Number List programmed for that time of the day by the System Engineer.
• If the Toll Control Level is 1, 2, 3 the outgoing call is allowed/denied as per the Call Privilege and the
corresponding number list programmed for each level.
• If Dynamic Lock - Automatic has been set by user/Operator, the system waits for the duration of the
Dynamic Lock Timer set for the extension. At the end of each outgoing call made during the period of this
Timer, the system will restart the Timer again. The system will change the Toll Control back to the pervious
Level when no call outgoing call is made till the expiry of this Timer.
• If Dynamic Lock - Automatic has been set by user/Operator, and an internal call is made during the period
of the Dynamic Lock Timer, the system will check for the 'Decrement Dynamic Lock Timer - Internal Calls'
feature in the Class of Service of allowed to the extension. If this feature is enabled, the system will start
the decrement of the Dynamic Lock Timer. The system will change the Toll Control back to the previous
level on the expiry of this Timer. However, if the 'Decrement Dynamic Lock Timer' feature is disabled in the
Class of Service, the system will reset the Toll Control as described in the previous step.
• If Dynamic Lock - Manual has been set, the extension user/Operator must set the Toll Control Level back
to the previous Level.
Feature Interactions
• Redial and Auto Redial: The system will check for Toll Control Level when an extension, on which
Dynamic Lock is set, attempts Redial or Auto Redial.
• Emergency Number Dialing: All extensions will be able to dial Emergency numbers always, regardless
of the Toll Control set on them.
ETERNITY provides for a separate programming of Emergency Numbers, which remain unaffected by
Dynamic Lock set on the phones. Refer the topic “Emergency Dialing” to know more about this feature.
How to configure
For this feature to work, it must be enabled in the Class of Service of the extensions; Toll Control Level must be
programmed in the Station Basic Feature Template of the extensions. The user must change the default User
Password.
Retain the default template, if you want to allow this feature to all extensions and keep the Decrement Timer
disabled.
If you want to deny Dynamic Lock to all extension, simply disable this feature in the default COS group 01 of Station
Basic Feature Template 01.
If you want to allow Dynamic Lock and or the Decrement Dynamic Lock Timer for Internal Calls only to selected
stations, then follow these steps:
a. Define a new CoS group with Dynamic Lock and the Decrement Dynamic Lock Timer for Internal Calls
enabled.
b. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
c. Assign this newly prepared Station Basic Feature Template to the extension on which 'Dynamic Lock' and
'Decrement Dynamic Lock Timer’ for Internal Call is to be allowed.
Similarly, if you want to deny Dynamic Lock/Decrement Dynamic Lock Timer to selected extension, prepare a new
Station Basic Feature Template with this feature disabled in the CoS group. Assign this feature to those stations
which are to be denied this feature.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for programming instructions.
How to use
Dynamic Lock, Manual and Automatic, can be set by extension users as well as from their own extensions or from
the SA mode by the Operator.
The extension user/Operator must first change the Dynamic Lock Level and set the Dynamic Lock Timer.
To set Dynamic Lock-Manual, the extension user/Operation must set the Dynamic Lock Timer to 00.
Recall that
• When the Dynamic Lock-Manual is set (Timer set to 00), the extension user/Operation must dial the
feature access code to restore the previous Toll Control Level.
• When Dynamic Lock-Automatic is set (Timer set to desired number of minutes), the system will restore
the previous Toll Control Level at the end of the Timer.
• The extension user must change the default User Password to be able to set the Dynamic Lock on his/
her extension. Refer the topic “User Password” for instructions on changing the password.
OR
• Dial 141.
OR
If you are still working with the default User Password, the system will prompt you to 'Change User
Password' when you attempt to set Dynamic Lock. Change your User Password first, before you use this
feature.
OR
• Dial 142.
OR
OR
OR
OR
• Dial 1072-002179.
OR
179. Ensure that the feature 'Allow SA commands' is enabled in the “Class of Service (COS)” allowed to the extension from which this
code is being dialed.
OR
• Dial 1072-002180.
OR
OR
• Dial 1072-002181.
OR
180. Ensure that the feature 'Allow SA commands' is enabled in the “Class of Service (COS)” allowed to the extension from which this
code is being dialed.
181. Ensure that the feature 'Allow SA commands' is enabled in the “Class of Service (COS)” allowed to the extension from which this
code is being dialed.
What’s this?
• E&M connectivity feature of ETERNITY offers seamless connectivity in PLCC network and also in between
various communication products like PBX, Router, Lease Line.
• E&M interface is widely used interface to connect such diverse equipment. For example, in a PLCC
network, number of PLCC EPAX needs to be connected. As shown in the figure 1 of PLCC network,
number of EPAXs are connected with each other through E&M tie lines.
• Say, an existing PBX capacity needs to be expanded beyond the configuration limit of a PBX. Installing
one more PBX and connecting both the PBXs through E&M interfaces can get us the desired expansion.
All E&M ports can be put together in one single group depending on the requirement. A separate access code
can be assigned to it. This makes the operations easy. Now the subscribers have at least two different access
codes for making outgoing calls using trunk lines and other for making outgoing calls using E&M tie lines.
PBX-A PBX-B
T1 E&M1 E&M1 T1
T2 E&M2 E&M2 T2
PSTN E&M3 E&M3
PSTN
Tn Tn
S1 S2 Sn S1 S2 Sn
• S1 to Sn are stations.
E&M1 E&M1
PBX-A PBX-B
E&M2 E&M2
T1 (8x8x8)
(16x8)
E&M3 E&M3 T1
E&M4 E&M4
T2 E&M5 E&M5
T2
PSTN E&M6 E&M6 PSTN
E&M7 E&M7
E&M8 E&M8
Tn Tn
S1 S2 S8 S1 S2 S16
PBX-C
(24x8)
S1 S2 S24
Figure 4: Two PBX systems located far from each other connected to each other using E&M connectivity.
PBX-A
T1 Router
PSTN T2 E&M E&M
T3
S1 Sn
Lease
circuit,
VSAT
2001 2099
PBX-B
T1 Router
PSTN T2 E&M E&M
T3
S1 Sn
2001 2099
How it works
All E&M ports can be put together in one single group or few groups depending on the requirement. A
separate access code can be assigned to it. This makes the operations easy. In such case the stations can
have at least two different access codes for making outgoing calls, one for making outgoing calls using trunk
lines and other for making outgoing calls using E&M lines. Generally, E&M connectivity is used to expand the
PBX capacity or connect two or more remotely located PBXs. This forms a network of PBXs. The requirement
is that, so formed network should work as one Group. This is commonly known as Closed User Group. Please
refer “Closed User Group (CUG)” and “Closed User Group-With Exchange ID” for more details.
How to configure
Please refer “Station Basic Feature Template”, “Station Advanced Feature Template”, “Trunk Feature Template”
for more details.
Relevant Topics:
1. “Station Basic Feature Template” 436
2. “Station Advanced Feature Template” 446
3. “Trunk Feature Template” 546
What’s this?
• The E1 Maintenance consists of Error Counts (Performance Statistics), Alarms and Loop Back Tests.
• G.775 is also considered for detection of defect conditions like Loss of Signal (LOS), Loss of Frame (LOF),
Alarm Indication Signal (AIS), etc.
• To elaborate, the Digital line can have transmission errors. All the errors will not generate an Alarm. Few
severe errors generate Alarms. However, all the errors are logged in the System Fault Log.
• The SNIIC (Subscriber Network Interface Integrated Circuit), is used to interface E1 line to ETERNITY. It
supports error counters listed in the table given below.
• Each error detected by the ETERNITY ME Card T1E1PRI/port is sent to the master in the form of an
event.
• The master counts these errors and prepare a statistical record if the condition matches. For example,
Severely Errored Seconds Count is incremented when one OOF (Out of Frame) event reaches the master
or more than 320 framing errors reach the master.
• FAS: The D channel can provide signaling for the other B channels on the same interface. This is
called ‘Facility Associated Signaling’ (FAS).
• NFAS: The D channel can provide signaling for the other B channels on more than one interface. This
is called ‘No facility Associated Signaling’ (NFAS). The signaling arrangements, the capability is
supported to designate a D channel on one interface to be a backup to a D channel on another
interface in case of failure. This is called D channel backup.
Errored Frame Alignment Signal This counter is incremented on receipt of each errored FAS.
E-bit This counter is incremented when either E1 or E2 bit is set in the transmit
frame.
CRC-4 Error This counter is incremented when the received frame has CRC-4 errors.
Line Code violation Error This counter is incremented when a line code violation error occurs.
Excessive Zeros Error This counter is incremented when excessive zeros are received or Line
code violation error occurs.
Positive Slip Buffer This counter is incremented every time a positive slip occurs.
Negative Slip Buffer This counter is incremented every time a negative slip occurs.
Following parameters form the statistical record. This can be generated in the form of a report as shown below:
Out Of Frame (OOF)-Out of Frame is the occurrence of a particular density of framing error events. OOF is
declared when three consecutive frame alignment signals have been received with an error. OOF ends when;
Severely Errored Framing Seconds (SEFS)-It is a second with either one or more OOF defects or a detected AIS
defect.
Unavailable Seconds-It is defined as a second in which E1 service is unavailable. An unavailable state is declared
at the onset of 10 consecutive severely errored seconds and is cleared on onset of 10 consecutive seconds with no
severely errored seconds.
Positive Slip Seconds-It is defined as a second in which a frame is repeated to account for frequency drift
between ET2 and the network.
Negative Slip Seconds-It is defined as a second in which a frame is deleted to account for frequency drift between
ET2 and the network.
Loss of frame count-Loss of Frame is declared after 2.5 seconds of continuous loss of signal or OOF. LOF is
cleared after 10 seconds of continuous no loss of signal or OOF.
Line Errored Seconds-It is a second in which one or more than one line code violation error occurs.
Excessive Zeroes Error Count-This counter is incremented when excessive zeroes are received on the line or
when line code violation error occurs.
Alarms
RED Alarm:
YELLOW Alarm:
If equipment is connected in downstream (Drop and insert mode i.e. NT mode) then on receipt of Yellow
Alarm, a Blue Alarm will be sent on the port, which is configured in NT mode.
BLUE Alarm:
L1 Red Flashing @1 Sec. Card Heart Bit Port 1 Layer is not established LOS-Red alarm detected.
L1 Red Flashing @500 Card Heart Bit Port 1 Layer is not established LBFA -Loss of Basic Frame
msec. Alignment.
L1 Red Flashing @100 Card Heart Bit Port 1 Layer is not established- LMFA loss of multiframe
msec. alignment.
L1 Yellow Flashing @1 Sec. Card Heart Bit Port 1 Layer is established -RAI (Yellow Alarm) is detected.
L2 Yellow flashing @1 Sec. Port 1 doesn’t achieves CRC4 synchronization-AIS (Blue Alarm)
is detected.
L3 Red Flashing @100 Port 2 Layer is not established-LMFA Loss of Multi Frame
msec. Alignment.
Relevant Topic:
1. “T1E1 Trunks” 1608
What’s this?
Emergency Conference enables you to establish a Conference between a pre-defined group of extensions using a
feature access code.
This feature can be used to call and consult with a group of people in emergency situations.
The number of parties that can be included in an Emergency Conference group depends on the Multiparty
Conference Capacity of your model of ETERNITY. The ETERNITY supports between 6 to 21 parties in a Multiparty
conference depending on the model you are using. For details, see “Conference-Multiparty”.
How it works
For this feature to work, you must do the following:
• First decide the key persons in the organization who should be parties to the Emergency Conference.
• Form a Department Group with the extensions of these key persons as members. A single Department
Group can have up to 32 extensions. For more information on forming Department Groups, see the topic
“Department Call”.
For example, you have formed a Department Group for Emergency Conference, with the extensions A to G as
members. The Access Code assigned to the Department Group is 3901.
Now, extension H wants to initiate an Emergency Conference. This is how the feature will work:
• All extensions in the Department Group (extensions A to G) which are free will start ringing. Extensions
that are busy will not be included in the call.
If there are DKP/Extended IP Phone extensions in the group, and these phones have a Call Appearance
free, the system will ring these extensions on the free Call Appearance, but will not wait for the extensions
to become free.
The number of extensions that the system will ring will depend on the resource occupied in the
conferencing circuit in the system at the time of initiation of the Emergency Conference. For example,
ETERNITY GE supports up to 15 participants in a single Multi-party conference, and 5 simultaneous
conferences, if all conferences involve 3 parties. Now, if there are already three such simultaneous
Multiparty conferences in the system when Emergency Conference is initiated, the system will ring on the
first 6 extensions of the Emergency Conference group, even if the group has more extension members.
This is because the system supports 15 participants and 9 parties are already involved in the three
simultaneous conferences.
• Two-way speech is established with extension A and H. All other extensions continue to ring.
• When another extension, B goes Off-Hook to answer the call, A and H get a beep, and three-way speech
is established between A, B, H.
• Thus, whenever a new member joins the conference, all other extensions already in conference will get a
beep, if the flag Play Beep when Conference/Dial-In Conference Starts is enabled in the “System
Parameters”.
• If the conference initiator, extension H, goes idle, all other extensions in the conference will still be in
conversation.
• When speech is established with one or more member extensions of the Emergency Conference
department group.
Or
• During Ring Back Tone, as the system rings on the extensions of the group, after the initiator of the
conference has dialed the feature access code.
• To cancel the Emergency Conference, extension H must dial the feature access code for Cancel
Conference, 190 (default).
How to configure
To provide this feature to extensions,
• You must enable the feature 'Emergency Conference' in the“Class of Service (COS)” of the extensions in
their “Station Basic Feature Template”. By default, this feature is enabled on all extensions, so all
extensions can use this feature.
• If the extension you are providing this feature is a DKP or an Extended IP phone, you may program a DSS
key on the phone with this feature.
• You must also create a Department Group as Emergency Conference group. For instructions, see
“Department Call”.
• By default, the system plays a beep when the Emergency Conference starts. If you do not want the beep
to be played, you must disable the flag Play Beep when Conference/Dial-In Conference Starts. For
instructions, see “System Parameters”.
This flag is common for other features like “Conference-Multiparty”, “Conference Dial-In”, “Emergency
Conference”, and “Raid”.
If no resources are free, you will get the ‘Conf. Resource full’ message on your LCD.
• Go OFF-Hook
• Dial 190
• All participants who are connected, get disconnected and get error tone. Extensions participants who have
not yet answered the call, will stop ringing.
You can cancel the conference only if you have initiated it.
• Dial 1177
• Dial Department Group Number.
• Go OFF-Hook
• Dial 190
What’s this?
When an emergency call is made from an extension, the system dials out the number using any of the free trunks
selected for routing Emergency Numbers. Since the number is dialed out by the PBX, the Emergency Service that
attends to the call will be able to locate the PBX, but not the extension that made the call.
Similarly, the Operator too has no way of knowing which of the extensions made the call, thus making it difficult to
quickly reach and provide help to the extension that made the emergency call.
With the Emergency Detection and Reporting feature, the Operator can know from which extension the emergency
call is being made. Whenever an Emergency call is made by an extension user, the system detects and reports it to
the Operator extension.
How it works
When an extension of ETERNITY makes an emergency call by dialing an Emergency Number,
• the system hunts for a free trunk in the OGTBG selected for routing the emergency number, and dials out
the number from a free trunk.
• simultaneously, the system informs the Operator by ringing on the Operator extensions for the duration of
the Alarm Ring Timer (configurable; default: 45 seconds).
• If Operator is a DKP or an Extended IP Phone, it will ring continuously, and an emergency message will be
displayed on the LCD.
The emergency message shows the number of the extension which has made the emergency call, in this
case, extension 2003.
Also see the topics “Configuring Emergency Number Dialing” and “Emergency Dialing”.
How to configure
To be able to use this feature, the Emergency Dialing Reporting flag must be enabled in the System Parameters.
See “System Parameters” for instructions. By default, this flag is enabled.
What's this?
The ETERNITY supports dialing of Emergency number immediately without any blocking.
When an extension user dials an Emergency number, the system will hunt for a free trunk from the outgoing trunk
bundle group selected for the emergency number. See “Configuring Emergency Number Dialing”.
The system will not apply any of the following on the extension dialing the Emergency number:
• Toll Control (Allowed Denied Numbers, Dynamic Lock)
• Call Budget (even when call budget is consumed)
• Call Duration Control
• Automatic Number Translation
The system will allow the extension to dial the Emergency number even in the following conditions:
• the extension is in Off-Hook state.
• the extension is in Standby Mode.
• the extension has grabbed the trunk line (using Trunk access code or selective access)
• the call state is in any state: Ringing, Busy, Error, Confirmation.
• SIM card is not present in the Mobile port.
• Mobile port is not registered with the network.
• SIM PIN is not valid.
• the keypad of the extension phone is locked.
Emergency Numbers will always be out dialed through the OGTBG you have selected for the numbers, except
when you have grabbed a trunk using Selective trunk access code/Selective Trunk Access DSS key. In which case,
the number will be dialed only from the trunk you have grabbed.
Emergency dialing will not work if Mains Power to the ETERNITY fails.
How to configure
The Emergency numbers are fixed as per the Region where ETERNITY is installed, you can add emergency
numbers, as required. For instructions, see “Configuring Emergency Number Dialing”.
How to use
To dial an Emergency number,
• Go Off-Hook
• Dial the Emergency Number
OR
• Dial Trunk Access Code-Emergency Number
• Wherever the Trunk Access Code conflicts with the Emergency Number, the emergency number
should be dialed after dialing the Trunk Access Code.
• Let us take the example of Australia, where the emergency number is 000 and the trunk access code is
0. Now, when an extension user of ETERNITY located in Australia dials ‘0’ of the emergency number,
the system will consider it as trunk access code and will apply the trunk access code logic.
• Therefore, in such cases, the extension user must first dial the Trunk Access Code and then the
Emergency Number. In this case, the extension user must dial 0-000 for emergency number dialing, so
that the system will not wait for the Conflict Timer to apply the Trunk access code logic.
What’s this?
ETERNITY provides a facility to give external music to the users by way of connecting an external music source. It
is possible to give external music to the stations or external callers or both. For more details, please refer “Music on
Hold (MOH)”.
Relevant Topic:
1. “Music on Hold (MOH)” 1335
What is Flash?
Pulse dialing is a type of signaling in which codes (digits) are dialed in pulses. A hook switch or a Flash key is
generally used to dial this code. Technically, Flash is breaking the loop current for 200 milliseconds to 900 ms.
Please note that since this code is not simulated in standard DTMF convention, one cannot dial it in DTMF mode.
Flash timer signifies the time period for which the loop current breaks. Flash timer is programmable. Flash timer
ranges from 083 ms to 999 ms. By default, Flash Timer is 600 ms.
Where is it used?
Stations dial flash to use few PBX features and also to use few PSTN features. Flash is used in following cases:
Now a days, more number of basic service providers and different types of advanced electronic telephone
exchanges are prevailing. It is possible that one service provider interprets breaking of loop current for 300 ms
as flash, other service provider interprets breaking of loop current for 900 ms as flash and system interprets
breaking of loop current for 600 ms as flash. Hence if the system engineer sets the flash timer to 600 ms then
he might not be able to use features provided by the service provider interpreting 900 ms for flash.
To take care of this situation, ETERNITY offers Flexibility to program different flash timers for both stations and
trunks.
How to configure
Step 1
Please refer the topic “SLT Hardware Template” for more details on assigning flash timer to a SLT.
Step 2
Please refer the topic “TWT Hardware Template” for more details on assigning Flash Timer to E&M.
Step 3
Please refer “E&M Feature Template” for more details on assigning Flash Timer to an E&M.
• Many times it happens that while transferring the call, the call either gets disconnected or is not
transferred. This happens due to mismatch of time for which the hook switch is pressed, if used for
transferring the call hence it is advisable to use Flash Key of the telephone instrument, instead of hook
switch.
• This problem may occur with Flash Key also, if the timer for the Flash Key on the telephone instrument
and the flash timer of the system are not set properly.
Relevant Topics:
1. “SLT Hardware Template” 428
2. “TWT Hardware Template” 530
3. “Digital Key Phone-Operation” 1123
4. “E&M Feature Template” 565
What’s this?
Trunk exchanges support many advanced features like call waiting, call forward. To use these features it is
required to dial codes during speech. The dialing of codes during speech do not create problem when you are
dialing on the trunk directly. But with the PBX connected between the user and the central office, the central office
codes clash with PBX codes. This leads to difficulty in accessing CO features while in speech. However,
ETERNITY supports dialing codes on trunk when in speech to all codes from any extension. But it is required to
inform the PBX, prior to dialing some code on the trunk.
How to use
Example:
To use Call Waiting facility of service provider exchange from any station, perform following steps:
How to configure
Please refer “Class of Service (COS)” for details on how to allow Flashing on Trunk to a user.
Relevant Topics:
1. “Class of Service (COS)” 1011
2. “Flash Timer” 1219
What’s this?
• ETERNITY offers Flexibility to assign a code of your choice to access a station. This code is called Flexible
number. For example, to access first SLT having software port 001, one has to dial 2001. It is possible to
change this code to any other number of your choice.
• ETERNITY offers two types of stations viz. SLT and DKP. The system loads default access codes to all the
SLT and DKP stations on first power ON. Later on the stations can be assigned default Flexible numbers
using a command.
001 2001
002 2002
003 2003
: :
512 2512
001 3001
002 3002
003 3003
: :
128 3128
How to configure
Use following command to program the access code for a SLT station:
3101-1-SLT-Access Code-#*
Where,
SLT is a 3-digit number from 001 to 512.
Access Code is maximum of 6 digits.
The flexible number can be programmed up to 6 digits, however the SE is recommended to program
flexible number of 4 or less than 4 digits only.
• It is possible to clear the flexible number of a station, range of station and all stations.
• Flexible numbers are the codes dialed from dial phase to call another station. These flexible numbers
should be unique and should not match with either other SLT stations or DKP stations or any of the
features available from the dial phase.
• Flexible number having common digits can be assigned to another station. Please refer “Conflict
Dialing” for more details.
• Use flexible numbers for all the features used from User mode and SA mode. Software port numbers
are to be used only during for SE mode.
• When the access code of a station is cleared; its flexible number becomes null or void.
• If access code of a station is cleared, one cannot call that station. However the station with NULL
flexible number can make calls as usual.
Relevant Topics:
“Conflict Dialing”
“Access Codes”
What's this?
The Floor Service feature allows you to provide a common access code to extension users which they can dial to
call floor service.
Essentially a hospitality feature, Floor Service is also useful in offices. Floor service can be any administration or
service department in the building, such as a stationery room, back office, backroom, photocopy/ mail room,
secretarial assistance, concierge/janitor, Storeroom.
Just as all extension users can reach the Operator by dialing the common access code '9', they can reach the floor
service by dialing a common access code, '38'. This is the default Floor Service access code, for all geographical
regions where ETERNITY is installed.
• Multi-storied buildings, which have floor service (pantry, mail sorting, house keeping, janitor, coffee room,
refreshment area) for each floor. The ETERNITY can be programmed to land calls made by extension
users dialing the common access code '38' on the floor service extensions of their respective floors.
• Offices that have a centralized floor service, instead of one on each floor. The ETERNITY can be
programmed to land calls made from all extension phones by dialing '38' on the common floor service
extensions.
How it works
For example, Midas Towers houses different departments on each floor. Each floor has Floor Service.
Extensions 2001 to 2010 are on the first floor, 2011 to 2020 on the second floor, and 3001 to 3010 on the third floor.
The floor service extensions are numbered as 2012 on the first floor, 2022 on the second floor and 3012 on the
third floor.
With the Floor Service programmed for each floor, when the extension user 2001 dials '38', the call will land on the
service extension 2012, assigned to room service on the first floor. Similarly, when the extension user 3008 on the
third floor dials '38', the call will land on the service extension 3012 on the third floor.
If Midas Towers had a single floor service extension 2012 for all floors, with Floor service programmed, calls made
from all extensions by dialing '38' would land on extension 2012 only.
How to configure
Programming the Floor Service feature involves the following steps:
1. Creating a routing group for each floor. Include Floor service extensions of a floor in a routing group
prepared for that floor.
2. Assigning a routing group (number) in the Floor Service feature in the Station Advanced Feature
Template. Prepare a different Station Advance Feature Template for each floor.
If the Enterprise/Building has centralized floor service, you only need to create a single Routing Group with
service extensions, as required. This routing group number can be programmed on a common Station
Advanced Feature Template which will be applied to all extensions.
• Choose the Routing Group number (01-96) you want to use as floor service group.
You can program different routing groups for different floors. In each routing group you can program
maximum 32 service extensions as 'members'.
• For routing group to be used as floor service, program the following parameters:
• Rotation Flag: With this flag, you can enable or disable the rotation of calls in the routing group which
has multiple 'member' extensions. When enabled, each fresh call will land on the extension which is
next to the one that received the last call. This ensures equal distribution of incoming calls to all the
• Member Type: Select the 'Member Type'. If the floor service extension is an SLT, select SLT; if it is a
DKP, select DKP as member type. Similarly, if the extension is an ISDN Terminal, select ISDN
Terminal.
Program only as many extensions as you want in the routing group and set the remaining Member
Types to 'None'.
For example: if you want to program only one extension in the routing group, set the Member Type in
the remaining columns (Member 02-Member 32) to 'None.'
• Port Number: Enter the software port number on which the SLT/DKP/SIP Extension/ISDN Terminal
floor service extension is attached.
• Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should ring.
By default, the ring timer is set to 015 seconds and can be changed.
• Continuous Ring Flag: With this flag, you can set an extension to ring continuously until the call is
answered. The first extension will continue to ring even as the system hunts for other extensions in the
routing group to land the call. If the call still remains unanswered, the system will return the call to the
first extension once again. This flag is of no relevance, if there is only one member extension in a
routing group.
• Repeat the above steps to include other floor service extensions in the routing group.
All extensions are assigned the Advanced Feature Template 01, by default. If the enterprise requires
separate floor-service group for each floor, program a separate Station Advanced Feature Template for the
extensions of each floor.
• Select a Station Advance Feature Template number to be assigned to the user extensions of a floor. For
example: Template number 02 for extensions 2001 to 2010 are on the first floor, Template number 03 for
• Scroll with the horizontal bar to reach the column 'Floor Service' of the selected Templates. Enter the
Routing Group number you want to use as floor service group for that particular Station Advance Feature
Template.
• Now, apply the Station Advanced Feature Templates (with floor service routing groups programmed) to the
extensions of the respective floors. For example: Template number 02 on extensions 2001 to 2010 are on
the first floor, Template number 03 on extensions, 2011 to 2020 on the second floor, and Template number
04 on extensions 3001 to 3010 on the third floor.
If extensions are SLT, assign the Template on the 'SLT Parameters' page.
If extensions are DKP, assign the Template on the 'DKP Parameters' page.
If extensions are ISDN Terminals, assign the Template on the 'ISDN Parameters' page.
Port Number is the Software port number182 on which the floor service member extension SLT, DKP,
ISDN Terminal is attached.
Software port number of the SLT, from 001 to 512.
Software port number of the DKP, from 001 to 128.
Software port number of the ISDN Terminal, from 01 to 64.
Software port number of the SIP Extension, from 001 to 999.
To program the Continuous Ring Flag for the routing group, dial:
• 6504-1-Routing Group-Destination Index-Flag
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is from 01 to 32
Continuous Ring Flag is
0 for disable continuous ring (each member extension in the group will ring for the programmed 'Ring
Timer' for the group)
1 for enable continuous ring (the first extension in the group will ring till the call is answered)
To apply the Station Advanced Feature Template now programmed with the Routing Group to SLTs,
DKPs, ISDN Terminals, SIP extensions, refer the topic “Customizing Station Advanced Feature
Template using a Telephone”.
182. Refer the topic 'Software Port and Hardware ID' in the ETERNITY System Manual.
183. Enter the number of the routing group you programmed as Floor Service group.
How to use
To be able to use floor service, extension users may dial the default Access Code defined for Floor Service: 38.
Check with your System Engineer if this access code has been changed and dial the new access code
obtained from the System Engineer.
OR
What’s this?
Using this feature, extension users can make your calls follow you wherever you go. Extension users can receive
their calls on another extension, whenever they want.
How it works
• A’s extension number is 2001.
• B’s extension number is 2003.
• A is currently at B’s extension.
• A wants to receive calls from extension 2001 on extension 2003.
• A sets Call Follow Me on extension 2003.
• All calls landing on A’s extension 2001 will be forwarded to extension 2003.
• When A returns to extension 2001, A cancels Call Follow Me.
• The extensions dial tone changes to feature tone if its calls are forwarded.
• Multiple users can use ‘Follow Me’ from the same extension.
• Follow Me can be overwritten. Extension A sets Follow-Me on extension B. After a period of time; goes
to extension C. A can receive calls on extension C by setting Follow Me on extension C. Follow Me set
by A on extension B will be cancelled.
• Follow Me cannot be chained. If extension A sets Follow Me to extension B. And extension B sets
Follow Me on extension C, Follow Me of extension A is automatically cancelled.
Also see “Call Forward”, “Class of Service (COS)” and “Do Not Disturb (DND)”
How to configure
To be able to use Follow Me, extension users must have Call Forward feature enabled in their Class of Service for
the time zone. For instructions, see “Class of Service (COS)” and “Station Basic Feature Template”.
How to use
For EON & Extended IP Phone Users
What’s this?
Extension users can force other extension users to answer their calls when there is no response from the called
extensions.
How it works
Forced Answer can be requested from an SLT, a DKP and from the Extended IP Phone (calling extension).
However, the called extension (being forced to answer) must be either a DKP or an Extended IP Phone.
Forced Answer can be used when the called extension is busy, but has free call loops (call appearances)
for calls to land. For example:
• Extension A (SLT) calls extension B (DKP).
• B is in speech with extension C, but there is a call appearance free on B’s DKP.
• Since B’s DKP has a free call appearance, A’s call lands on B.
• B’s extension starts ringing.
• A dials Forced Answer code during ring back tone.
• Extension C is put on hold.
• A is in speech with B.
• A may now talk to B.
• When A disconnects the call, B is now in speech with C.
How to configure
To be able to use Forced Answer, extension users must have this feature enabled in their Class of Service for the
time zone. For instructions, see “Class of Service (COS)” and “Station Basic Feature Template”.
How to use
For EON & Extended IP Phone Users
You can also dial ‘5’, the feature code for Forced Answer, immediately after dialing the desired extension
number, instead of dialing it during Ring Back Tone. This way, you can talk to the desired extension user
without waiting for the called extension user to answer your call.
What’s this?
Forced Call Disconnection enables extension users to disconnect a busy extension or a trunk at will, and free the
system resources (access to extension and trunk) for themselves.
How it works
Forced Call Disconnection of an Extension:
• A, B and C are extensions.
• A and B are in speech.
• C calls B and finds it busy.
• C uses Forced Call Disconnection by dialing the feature command.
• C gets confirmation tone, while A and B get error tone.
To be able to use Forced Call Disconnection, the extension user must have a higher “Priority” than the
extension user whom he/she tries to forcibly disconnect.
To be able to use Forced Call Disconnection on a busy trunk, the extension user must have grabbed that
trunk using “Selective Port Access”. If the extension user has grabbed the trunk using a Trunk Access
Code, the feature code to dial Forced Call Disconnection will not work.
In PLCC applications, Forced Call Disconnection can be used in a chain to reach the last Exchange
through many tandem exchanges in between.
.
You are advised to restrict access to this feature only to important extension users. Extension Users who
are allowed this feature are advised to use it judiciously.
How to configure
To be able to use Forced Call Disconnection, the extension must have:
• Forced Release feature enabled in the Class of Service. For instructions see “Class of Service (COS)”
and “Station Basic Feature Template”.
• As Forced Call Disconnection on a busy trunk is possible only if the extension user has grabbed that trunk
using Selective Port Access, this feature must be enabled in the Class of Service of the extension. For
instructions see “Class of Service (COS)” and “Station Basic Feature Template”.
How to use
For EON & Extended IP Phone Users
• Press the DSS Key assigned to Forced Call Disconnection on Busy tone.
OR
184. Only if you have grabbed this trunk using Selective Port Access.
What's this?
• When ETERNITY acts as a Gateway, this feature is used to convey the call maturity information on source
port, when call made using destination port gets matured.
• This information can be useful for billing equipment connected at calling party side.
How it works
• This feature of answer signaling is applicable only for DISA option selected as 'CLI Authentication-one call-
Ans. Sig.' in the DISA-CLI Authentication Table. Refer chapter “Direct Inward System Access (DISA)”.
• The ETERNITY works as gateway for routing the call to the destination.
• If the calling party's number is programmed in Table - 'DISA-CLI Authentication', the ETERNITY will
consider the calling party as successfully logged in as station which is programmed as "Auto Login
station". The call will get answered by the ETERNITY. The caller will get dial tone.
• When called party answers the call (i.e. call on destination port gets matured), the Answer Signaling will be
done on source port, if enabled.
• Answer Signaling will be done in form of DTMF digit string as programmed on the source port.
• If Answer Signaling is disabled on the source port, the DTMF digit string will not be dialed out, even if it is
programmed.
• On receiving these digits, Billing equipment/PBX with which the calling party is connected, can consider
the call is matured and start billing.
How to configure
To configure 'Gateway Application-Answer Signaling' flag and 'DTMF String' on desired trunk refer following
chapters:
Mumbai Delhi
Kolkata Chennai
• GFX11s are configured for multi stage dialing, in which when ever caller dials the number using pay
phone, GFX11 will store the number, it will first make a call to ETERNITY's T1E1 line on which DISA - 'CLI
Auth- One Call - Ans. Sig.' is enabled.
• GFX11's SIM Numbers are programmed in 'DISA - CLI Auth.' table in ETERNITY.
• ETERNITY compares the Calling Number received on T1E1 Port with DISA-CLI Auth. Table.
• As the number is programmed in the table, ETERNITY answers the call and offers the trunk assigned in
Station Basic Feature Template of the station used as 'Auto Login'.
• When ETERNITY answers the call, the GFX11 sends the stored called number (which is actually dialed by
the caller) in DTMF digits.
• ETERNITY routes the call on this number using the offered trunk.
• When called party answers the call, ETERNITY will send 'DTMF Digit Strings' programmed on the T1E1
Port (Source Port, on which call originated) as a Gateway Answer Signaling'.
• The Pay Phone connected with FXS Port of the GFX11 is also configured to understand the same DTMF
string as call maturity.
• Pay Phone will start billing only on receipt of the desired DTMF digits.
Relevant Topics:
1. “Direct Inward System Access (DISA)” 1156
2. “TWT Hardware Template” 530
3. “E&M Feature Template” 565
4. “Configuring Mobile Trunks” 595
5. “T1E1 Trunks” 1608
6. “ISDN-BRI” 1263
7. “Configuring SIP Trunks” 635
What’s this?
GPAX application is one of the applications provided by ETERNITY and used in commercial establishment/Society/
Organization, etc. Group PBX are installed, operated and maintained by organizations/agencies. The owner will be
treated as the main hirer. The private exchange (PSTN)/service provider will provide junctions to the PBX and the
owner will pay the rental of the junctions, and the call charges. The number of junctions provided to the PBX should
be adequate to carry the traffic. The owner/agencies of the PBXs will be responsible for the payment of all charges
to the private exchange/service provider.
In this application say a PBX ‘A’ is connected to PSTN. PBX ‘A’ is given an exchange ID (say 2837). A number of
stations (say 001-100) can be connected to PBX ‘A’. When a station 015 dials 2837025, PBX ‘A’ interpret this
number to be dialed for the same system. However, for dialing a station number belonging to same system, it is not
necessary to prefix the station number with exchange ID. When a station 020 dials 2834537, PBX ‘A’ does not
interpret this to be dialed for the same system and hence will dial the digits on the trunk.
When a station user picks up the handset and dials any digit except the one programmed in the routing table will be
dialed on the trunk. If the user dials digit that is programmed in the routing table with Self-flag enabled, the system
will not dial the digits on the trunk since it would interpret these to be dialed for the same system.
For dialing the digits on the trunk it is required to program the routing table carefully. Route code should be
specified in route code column of routing table in association with Self Route Flag disable. This will make the call to
be routed on trunk which is specified in OG Trunk Bundle Group. Regarding this please refer “Closed User Group
(CUG)” and “Closed User Group-With Exchange ID”.
PBX A
T1
T2
PSTN 2837
Tn
S1 S2 Sn
In order to make internal calls, user is advised to program Routing Table such that one of the entries in routing table
should have route code as ‘#’, Strip digit count as ‘1’ and Self route flag as ‘1’ (enable). Doing so, when user picks
up handset and dials the required number with prefix ‘#’, the system interpret this number as an activity for the
internal users and waits for relevant access code.
Because of Strip digit count=1, the first digit dialed by the user (i.e. # in this case) will be ignored and next digit will
be processed which could be a feature code or a station number.
In GPAX application, Answer Signaling on SLT and Answer Supervision on trunk shall e programmed properly. Also
the Disconnect Supervision parameter shall be programmed as appropriate for proper billing of calls.
Following figures shows how call is established between users using PSTN and ETERNITY-GPAX.
PSTN
Customer Site
C Makarpura Alkapuri
Subscriber
Card
ETERNITY-GPAX
FXO Subscriber B
(Trunk) Card
Inter Exchange
Trunk interface Inter Exchange
FXS card Trunk interface
(SLT) card
Inter
Exchange
Trunk card
1. Call from A to B is shown by
2. Call from B to A is shown by D
C
ETERNITY-GPAX
Trunk
Subscriber
Card
SLT
Card
Inter
A Exchange
Trunk
interface card
MFC Operator
C
ETERNITY-GPAX
Trunk
Subscriber
Card
SLT
Card
Inter
Exchange
A Trunk
interface
card
MFC Operator
Relevant Topic:
1. “Closed User Group (CUG)” 1028
What’s this?
In commercial establishments/large societies/organizations etc. staff make calls from their rooms. It is required that
the cost of these calls is calculated so that amount of calls made by a staff member can be paid. GPAX provides a
facility which if enabled can calculate cost of each call if programmed properly.
To calculate the total cost of a call please refer following topic for more details:
“Call Cost Calculation (CCC)”, “Call Duration Control (CDC)”, “Call Budget”, and GPAX charge Internal Calls in
“Station Advanced Feature Template”.
In order to make billing for internal calls between GPAX users, GPAX charge internal calls flag to be set to enable
(please refer “Station Advanced Feature Template” for more details).
If GPAX charge internal calls flag is enabled, this call will be recorded in the Station message detail recording-
outgoing buffer. If GPAX charge internal calls flag is set to disable, the call made to an internal station will not be
billed and will be recorded in the SMDR-Internal buffer as normal internal call.
• GPAX has a dedicated memory space (commonly called buffer) to store details of each call.
• These calls are retained in the buffer even during power failure.
• Various reports can be routed either on the printer or on the computer from this buffer.
• Once the buffer is 100% full, the new call overwrites the oldest one.
• It is recommended that printing of various reports should be regularised on fixed dates. This should be
done regardless of whether the buffer is full or not.
• This will prevent spilling and subsequent loss of data. You can enable or disable call logging for individual
trunk.
• This also prevents frequent spilling of the buffer, as new local calls will not be recorded.
What’s this?
• In office/organization, it is required that the flexible numbers are given in such a manner that it identifies
the departments/blocks i.e. say subscribers for marketing department may start with ‘5’, for technical
support with ‘6’ and so on.
• ETERNITY offers Flexibility to the user to assign a code of your choice to access a station. This code is
called Flexible number. For example, to access first SLT having software port 001, one has to dial 2001. It
is possible to change this code to any other number of your choice.
• ETERNITY offers two types of stations viz. SLT and DKP. The system loads default access codes to all the
SLT and DKP stations on first power ON. Later on the stations can be assigned default Flexible numbers
using a command.
001 2001
002 2002
003 2003
: :
: :
512 2512
001 3001
002 3002
003 3003
: :
: :
128 3128
How to configure
Please refer topic “Flexible Numbers” for more details.
Dynamic Toll Control Level Number List can be programmed and assigned to:
Relevant Topics:
1. “Conflict Dialing”
2. “Access Codes” 750
3. “Flexible Numbers” 1222
4. “Dynamic Lock” 1195
What’s this?
An organization may have a Centralised Information Office which provides information related to different
departments such as HR, IT, or General information. For each department in the organization, an extension number
can be defined as a Help Desk.
How it works
• Extension 2002 is defined as Help Desk for HR policies and general rules.
• Extension 2016 calls the Help Desk extension 2002.
• If the Help Desk extension is busy, an Auto Callback request is set automatically on the Help Desk
extension.
• As soon as the Help Desk extension is free, the system will serve the auto callback request.
• The Help Desk extension calls back extension 2016.
How to configure
You can define an station as ‘Help Desk’ by enabling the ‘Help Desk’ flag in its “Station Advanced Feature
Template”.
What’s this?
Hot Desking enables extension users to use all the properties of their own extension from another extension.
Hot Desking is useful for people who are often away from their own desks and must work from another. Hot
Desking allows them to use all the features and facilities of their own extension from another.
How it works
This feature is supported on DKP and SLT extensions only.
Hot Desking is possible only between extensions of the same type: SLT to SLT and DKP to DKP extensions.
The User Password of both extensions involved in Hot Desking must not be 1111.
Hot Desking can be performed only when both the extensions are idle.
• When Hot Desk is performed from the Hot Desking extension, all the properties of the Host Extension are
copied to the Hot Desk Extension.
• On the Host Extension, the user cannot perform any activity except Cancel Hot Desking.
• You must cancel Hot Desk from both the Hot Desk Extension and the Host Extension.
• After cancelling Hot Desk, the Host Extension and the Hot Desk Extension acquire their original properties.
How to configure
For this feature to work, the feature 'Hot Desk' must be enabled in the Class of Service of the Host Extension and
the Hot Desk Extension. See “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
How to use
The User password of both the extensions involved cannot be default password.
• On the Hot Desk Extension, press DSS Key assigned to Hot Desk.
OR
• Dial 1091
• Enter own extension number (Hot Desk Extension Number)
• Enter own extension User Password
• You get confirmation, Hot Desk cleared
• Go ON-Hook.
• Go to the SLT extension (Hot Desk Extension) with which you want to swap your SLT extension (Host
Extension) properties.
• Lift the handset of the Hot Desk extension.
• Dial 1091
• Dial Host Extension number
• Dial Host Extension User Password
• Replace handset.
What’s this?
The Hotline feature connects the extension user immediately to a particular number or trunk, whenever the
extension user goes OFF-Hook.
You can set Hotline to connect immediately to another extension, to a Department Group, to an external number or
to an outgoing trunk.
Hotline set for external numbers and outgoing trunks is referred to as Hot Outward Dailing.
• Delayed: When the extension user goes OFF-Hook, the system plays Dial Tone to the extension user and
waits for the Hotline Timer (default: 3 seconds). On the expiry of this timer, it connects the extension user
to the desired hotline extension number, department group, external number or outgoing trunk.
How it works
• Hotline/Hot Outward Dialing can be set from an SLT, DKP or Extended IP Phone extension, if the
extension has Hotline in its Class of Service.
• To be able to use Hotline/Hot Outward Dialing, extension users must do the following:
• Select the type of Hotline they want to set on their extension; whether to an internal Extension Number,
a Department Group, or an External Number or Outgoing Trunk.
• Configure the Hotline Timer. For Immediate Hotline, extension users must set the Hotline Timer to ‘00’
seconds. For Delayed Hotline, extension users can set the Timer as per their requirement.
A frequently dials the number of B. So, A sets Hotline for B’s number and also sets the Hotline Timer to 5
seconds (Delayed Hotline).
• A goes Off-Hook
• ETERNITY plays dial tone and waits for 5 seconds
• If A dials a number within the Hotline Timer, ETERNITY outdials the number dialed by A.
• If A does not dial any digit within this time, ETERNITY dials B’s number.
• A gets connected to B.
If ‘Dial Tone’ timer of the system is less than Hotline Timer, the Hotline Timer will override the ‘Dial Tone’
timer. To know more about these timers, see “System Timers and Counts”.
• If A sets delayed Hot Outward Dialing for a Trunk or an External Number, the system will play dial tone
to A and wait for the duration of the Hotline Timer for A to dial digits. If A does not dial any digits within
this timer, the system connects A to the Trunk/External Number.
• If A sets immediate Hot Outward Dialing (Hotline Timer set to ‘00’ seconds), A will be connected to the
Trunk/External number as soon as A goes Off-Hook.
• Delayed Hotline/Hot Outward Dialing allows extension users to dial out other numbers or grab another
trunk, without having to cancel the Hotline/Hot Outward Dialing they have set for a particular number or
trunk.
How to configure
To be able to use Hotline, extension users must have this feature enabled in their “Class of Service (COS)” for the
time zone, as required.
How to use
For EON & Extended IP Phone Users
You cannot set Hotline and Hot Outward Dialing on the same extension at the same time.
The cancellation code must be dialed from the dial tone. You have to be very quick in dialing the
cancellation code, if the delay in the Hotline Timer is set to 1 or 2 seconds.
When you set the Hotline Timer to ‘00’ seconds (for immediate Hotline), you will not be able to dial any
digits, not even the feature code to Cancel Hotline.
If you have set Immediate Hot Outward Dialing for a Trunk or External Number, you will not be allowed to dial any
feature code, not even the feature code to cancel Hot Outward Dialing. However, if you need to cancel, you must
follow the steps described below.
• Go OFF-Hook.
You get the CO network Dial Tone.
• Dial by Digit.
You will hear Pause/Silence.
• Dial the code to change the Hot Outward Dialing Timer (154) and change the duration of the timer
–or–
Dial the access code to cancel the Hot Outward Dialing (150).
• Go ON-hook.
You get the return ring of the trunk.
• Go OFF-Hook again.
You get connected to the held trunk.
• Go ON-Hook.
What’s this?
The IC Reference Table is a set of general features that define the logic of resolving an incoming call and placing
on the target DDI station. An IC Reference Table is assigned to every port. This table in conjunction with DDI
Routing Table identifies the target station. The ETERNITY offers 64 such table each of which can be programmed
as per the requirement.
How it works
The IC Reference Table consists of the following parameters:
• IC Reference ID-This is the reference number acts as an identifier to the translation logic programmed in
the IC Reference Table. Any number of table can have the same reference number. An IC Reference ID is
assigned to ISDN and SIP trunks. When a call lands on the ISDN trunk, the system checks the IC
Reference Number assigned to it and identifies the corresponding DDI Routing Table for call resolving. For
more details on the complete Translation Logic please refer the topics “Direct Dialing-In (DDI)” and “DDI
Routing Table”.
• Start Channel Number-This is the First Channel Number for the trunk to which the logic is applicable.
• Total Channel Count-The Total number of channels of the trunk to which the IC Reference Table is
applicable.
• DDI Routing Reference ID-This is the DDI Routing Table's Reference number used by the IC Reference
table for mapping the received DDI number to a flexible number. It is a link parameter.
• Route on First Destination-This flag can be enabled or disabled. Once the station is identified, the
system checks the DDI IC routing flag of the station. If the flag is enabled, the call always lands on first
station of the MSN number. If the flag is disabled, the call is routed to the identified target station.
• Ring Timer-This timer signifies the time for which the station on which the incoming call is received rings.
On expiry of this timer if the Call is not answered the call is routed as per the programmed logic. Route to
TLG-When No Reply-When the DDI station does not answer the call, the system checks for this flag. If it is
enabled, then the system routes the call to the TLG assigned to the Trunk. It is in seconds.
• When No Reply-If the DDI station does not reply, the system checks parameter for processing the call
further. As per the option selected, the system does one of the following:
• Disconnect the call
• Route the call to Trunk Landing Group
• Answer the call automatically, greet the caller with a voice message, and on completion of message
disconnect the call.
• Answer the call, greet the caller with a voice message, and on completion of the message route the call
to Trunk Landing Group.
• Route the call to Voice Mail; to the DDI station’s Mail Box.
For this, a mail box must be assigned to the station. If the station is not assigned mail box, the caller will
hear the welcome message of the VMS, but will not be able to access the mail box.
• Trunk Feature Template - A Trunk feature template is assigned to each DDI Routing Table. This enables
the user to allow Auto Answer Time Zone wise, allow DID on a few numbers according to the time zone.
For more details refer the topic “Trunk Feature Template”.
How to configure
See the table at the end of this topic for Feature numbers and the codes.
• Exit SE mode.
Feature No. 01 02 03 04 05 06 07 08 09
: : : : : : : : : :
Parameters Value:
Code 0 00- 01-30 00-30 00-99 Disable 001-255 Disconnect Disconnect 01-50
99
4 Route to Route to
Voice Mail Voice Mail
What’s this?
For PBX users in countries, where the Calling Line Identification (CLI) received must be suitably modified before it
can be used to dial out the number, ETERNITY offers the feature ‘Incoming CLI Modification’.
The Incoming CLI received with the Country or Area Code, or both. However, the dialing pattern of the public
network may require the received CLI to be prefixed with additional digits, to dial out the same number. Or the
dialing pattern of the public network may require the CLI to be stripped off the prefixed digits to dial out the same
number.
With the feature ‘Incoming CLI Modification’ programmed, the ETERNITY detects whether the incoming CLI is a
local number, a national, or an international number. It modifies the incoming CLI accordingly, by adding or stripping
off the prefixed digits so that the number can be dialed out as per the dialing pattern supported by the public
network.
The modified CLI is presented to the extension phones and is stored in the “Call Logs”, and SMDR (see “Station
Message Detail Recording”). Extension users can call a number in the Call Logs without having to modify the CLI
manually.
How it works
• Incoming CLI Modification parameters must be programmed in the system considering the dialing pattern
supported by the local public network.
• Accordingly, ETERNITY matches the CLI received with the programmed parameters.
• It detects whether it is an international, national or local number.
• It modifies the CLI according as per the Modification parameters programmed.
• It presents the modified CLI to the extension; stores the modified CLI in the SMDR and in the Call Logs of
the extension, provided it is a digital key phone.
• When the received CLI is dialed out by the extension user from Call Log, ETERNITY dials out the same
number.
How to configure
For this feature to work, the parameter ‘Incoming CLI Modification’ must be programmed in the ‘System
Parameters’ of ETERNITY. This can be done from Jeeves or by dialing SE commands from a Telephone.
• Enable Incoming CLI Modification: Enable this flag if you want to use the Incoming CLI Modification
feature. By default, this flag is disabled.
If you receive CLI in dialable format, there is no need to use this feature. In such case, keep the flag
disabled. You do not need to program any of the CLI modification parameters.
• Country Code: Enter the Country Code of the country where ETERNITY is installed. The Country Code
helps ETERNITY detect whether the Incoming CLI received is a national or an international number. Do
not enter any prefix for the Country Code. For example, if your ETERNITY is installed in USA, enter only ‘1’
as the Country Code. Do not enter ‘+’ or “00’ as prefix to the country code ‘1’. By default the Country Code
is ‘91’ (India).
• Area Code: Enter the Area Code of the place where the ETERNITY is installed. The Area Code helps
ETERNITY detect whether the Incoming CLI received is a local number. Do not enter any prefix for the
Area Code. For example, if you want to enter Area Code for Mumbai, enter only ‘22’. Do not enter the
prefix ‘0’ to the area code. By default, Area Code is ‘265’ (Vadodara city).
• Prefix required for International Calls: Enter the digits that are required as Prefix for dialing International
Numbers. The prefix may be up to 5 digits, with numbers from 00000 to 99999. By default, ‘00’ is set as the
prefix for dialing International numbers.
• Prefix required for National Calls: Enter the digits that are required as Prefix for dialing long distance,
National (within the country) numbers. The prefix may be up to 5 digits, with numbers from 00000 to
99999. By default, ‘0’ is set as prefix for dialing national numbers.
• Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose from the following options:
• No (Area Code not required): select this option if your public telephone network does not require the
dialing of Area Code for local numbers.
• Yes (Area Code with Prefix required): select this option if you public telephone network requires you
to dial Area Code with a particular Prefix for local numbers. If you select this option, you must also
program the Prefix digits for the Area Code.
• Prefix digits to Area Code for local calls: If you have enabled the ‘Area Code required to make local
calls’ flag in the previous parameter, enter the prefix digits for the area code for local calls in this field.
Program the Country Code without any prefix. For example, if you want to program USA as Country
Code, enter ‘1’ only (without the prefix ‘00’ or ‘+’)
Program the Area Code without any prefix. For example, if you want to program Mumbai as Area Code,
enter ‘22’ only (without the prefix ‘0’)
• Exit SE mode.
What’s this?
• Using this feature, the operator will be able to allow/restrict internal calls. The operator has flexibility to
allow calls during day time and restrict calls during night time. In ETERNITY this feature is implemented by
following ways:
• Sometimes, a group of users occupy multiple rooms in the Enterprise. In such cases members of the
group would like to communicate only among themselves and service stations.
• Certain service stations should be able to dial any other service station or any guest (for example Nurses,
Operator).
• Certain stations should be able to dial only service stations (for example: patients, single user in the office).
• A group of station users need to dial amongst each other and the service stations (for example: doctors).
• Solo Group (for example patients, single user of the station). This is classified in group 00.
• Universal Group (Nurses and other services staff for example). This is classified in group 99.
• Friend's Group (for example, Group of visitors). Such a group of station users can be assigned to any
group number from 01 to 98.
From SA mode:
Dial, 1072-904-Flexible Number-Guest Group
OR
Press DSS key for 'Guest Group'
Where,
Flexible Number is the code given to access a station.
Guest Group is from 00 to 99.
By default, the guest group assigned is 99.
• The user can call any body in his 'guest group' number '99'.
• The station users in one 'guest group', '99' can call each other within the group as well as the station users
in other groups.
How to use
From SA mode:
Dial, 1072-045-Code
OR
Press DSS key for 'Enable/Disable Internal Call Block'.
Where,
Code Meaning
• When this feature is enabled, the LED on the DSS key assigned to 'Call Block' will glow Red and when
disabled, the LED will be turned Off.
• When the operator issues "Call Block”- Enable command, all the station users will be assigned guest
group = 00.
• When the operator issues "Call Block”- Disable command, all the station users will be assigned guest
group = 99.
• Even after issuing SA command for 'Call Block', the operator can use the SA command 1072-904 to
change the guest group of any station user.
• Please refer separate manual for more details about Hotel Applications for this feature.
What’s this?
Interrupt Request allows you to break into an on-going conversation after intimating the extension user about the
interruption.
In case of an important or urgent trunk call the operator can put the call on hold, interrupt the busy extension user to
inform about the urgent call and then transfer the urgent call.
How it works
• A, B and C are extension users.
• C calls A.
• C gets Ring Back tone (RBT) and A gets beeps indicating a new call.
• To answer C’s call, A must dial Flash before the expiry of the Interrupt Request Timer. A will be in speech
with C. B will be put on hold and will get music on hold.
• If A does not dial Flash before expiry of the Interrupt Request Timer, C’s call will be disconnected.
• After the conversation between C and A is over, when C goes on-hook, speech between B and A will be
re-established.
Feature Interactions
• Call States:
• Interrupt Request works only if the dialed extension is busy. The dialed extension may be busy with
another extension or trunk (external number).
• Interrupt Request works only if the user about to be interrupted in is in a two-way normal speech with
another user or external party.
• It will not work if the busy signal is due to the user being Off-hook, or in the middle of dialing, or
accessing a feature of the ETERNITY.
• “Call Toggle”: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
• “Priority”: No Interaction with Interrupt Request. If 'A' has lower priority than 'B' but has Interrupt Request
enabled; A can interrupt B.
How to configure
To be able to use Interrupt Request, extension users must have this feature enabled in their “Class of Service
(COS)” in their “Station Basic Feature Template”.
If required you may also change the default value of the Interrupt Request Timer. For instructions see “System
Timers and Counts”.
How to use
What’s this?
• BRI port of the ETERNITY configured for NT mode can be connected to the BRI Port of another PBX
configured for TE mode. In such case, the ETERNITY will behave as Transit Exchange.
• Dialing method (en bloc/overlap) on the BRI port is programmable. This is because many PSTN
exchanges support only Overlap receiving. Hence, in such cases the BRI port (configured for TE mode)
will send the called party number in overlap mode.
• ETERNITY supports a software port entity called "ISDN Terminal" which can be connected to the BRI-NT
port and will be treated as stations of the PBX.
How it works
• The ISDN numbering plan with ETERNITY revolves around Multiple Subscriber Number (MSN) and Direct
Dialing In (DDI).
• With MSN feature, practically all the station users or other specific terminals can be given a unique
telephone number. One station can be assigned a number 2765400, other station can be assigned
2765401, etc. Hence, for one subscriber the Service Provider can assign multiple numbers. Hence this
feature is called Multiple Subscriber Number.
• The ETERNITY can be programmed to land all the calls coming through various channels of the BRI line
on the Operator just like in case of normal trunk.
• Alternatively, using DDI feature of ISDN the calls can be made to land directly on the desired stations. To
accomplish this requirement, each station should be given a unique directory number. On assigning
directory number, a table is formed internally called DDI table as shown below:
000 03
005 04
006 05
008 06
009 07
• When a caller calls a MSN number, the call lands on the PBX. The PBX compares the incoming number
with the DDI table. If the incoming number matches with any number in the DDI table, it routes the call to
the specific station. If the incoming number does not match with any number in the DDI table then it is
matched with CLI number. If it matches with any number in the CLI table, the call is routed according to the
CLI table. If the number does not match with either of these tables then the call is routed to the landing
destination.
ISDN
Network NT 1 ETERNITY
Power
U-Interface S/T
(2 wire) Interface
• Most of the Service Providers provide the NT1 along with the BRI line.
• At the Customer's Premises, the BRI line is terminated on the NT1. The S/T interface of the NT1 is
connected to BRI port of the ETERNITY.
4 Tx
5 Rx
The configuration details of S/T interface (RJ-45) on NT1 are given below:
3 Rx1
4 Tx1
5 Tx2
6 Rx2
• Since 32 BRI ports are supported at present and as per protocol a maximum of 8 terminals can be
connected to the BRI port; at present 64 software ports are supported for ISDN terminals, namely 01 to 64
(Maximum 64 Terminals).
• ISDN terminals (software ports) do not have any hardware slot and port Id of their own.
• ISDN terminals (software ports) are associated to the BRI software port and the BRI software port has a
hardware slot and port ID of its own.
• Each ISDN terminal is assigned an access code (flexible number), station basic template, station
advanced template and a CPU group.
• When call is made from the ISDN terminal the BRI port will check the calling party number sent by the
ISDN Terminal. If the calling party number = Programmed access code of the ISDN Terminal which are
assigned to the BRI port, the BRI Terminal will process the call further as per the Station Basic Feature
Template and Station Advanced Feature Template assigned to the ISDN Terminal.
• If the ISDN terminal doesn't send the calling party number while making the OG call, the Station Basic/
Advanced Feature template assigned to the BRI port will be used, for call processing.
• When the ISDN terminal doesn't send the calling party number = Access code programmed for the BRI
port on which it is connected, while making the OG call, the Station Basic/Advanced Feature Template
assigned to the BRI port will be applied.
• When ISDN terminal sends calling party number which is not programmed for any ISDN terminals
assigned to that particular BRI Port, neither its access code of that particular BRI port, the call will be
dropped, by the BRI port.
• When the BRI port is connected to a Public ISDN exchange, it behaves as a terminal. ETERNITY supports
this function of BRI Access by assigning it a parameter viz. Orientation = Terminal. The signaling used for
this Orientation is Q.931-protocol.
• When the BRI port is connected to an ISDN Phone or ISDN Video phone, it behaves as a 'network'.
ETERNITY supports this function of BRI Access by assigning it a parameter viz. Orientation = Network. In
this case, the BRI port behaves as network. The signaling used for this Orientation is Q.931- protocol.
• When the BRI port is connected to a private ISDN (the main application of this configuration is CUG,
feature transparency, etc.), the BRI port is used as a pipe of 128Kbps. It is of no significance which end
acts as network and which acts as terminal (User) since the role of the BRI port in such case will be to
route the call depending on the signaling protocol applied on the BRI port (128Kbps link). ETERNITY
supports this function of BRI Access by assigning it a parameter viz. Orientation = Tie-line. In this case, the
BRI port behaves as 128Kbps link. The signaling used for this Orientation can be any of the Inter-
exchange signaling protocol for BRI Access. The most commonly used is QSIG.
• Video Phone
• Making Data Call
Video Phone
• An important application of BRI-NT is establishing a Video Call using Video phone. It is used just like other
ISDN Phones.
• Video Phone can be used with ETERNITY supporting a BRI connection. ETERNITY does not support call
transfer from one Video Terminal to another Terminal. But the call routing is implemented by preparing
suitable OG Trunk Bundle Group (OGTBG).
• The OGTBG for the Video Phone should be so formed by the SE that a BRI channel will be allotted by the
Call Processing logic (preferably).
• If the OGTBG formed by the SE contains a non-ISDN trunk and if by the OGTBG logic this non-ISDN trunk
is to be allotted to the ISDN phone which has requested 2 B-channels then the call will be dropped. The
non-ISDN trunk is allowed only if the ISDN user makes an audio call. The PBX will allot first channel even
if other channels are available.
• The ETERNITY does not support logic for converting audio call to video call.
• Instead, the Video Phone user will have to use one Trunk Access Code (and hence OGTBG) to make and
receive Video call and another TAC to make an audio call.
• Depending upon terminal equipment call, the call will be answered by the Operator and transferred to the
Video Phone.
• This feature works only if it is supported by the Service Provider. Video Conference is established mainly
by the Video Conferencing (VC) equipment.
• A Video Conference system (H.320) can be connected to any of the BRI-NT port.
• For Video conferencing, three BRI-NT ports of ETERNITY are connected to the three ports of the VC
equipment (The remaining one port of the Video Conference system remains free).
• Thus user will assign at least 6 B-channels to the OGTBG that is to be assigned to the VC equipment.
(User can assign BRI-NT software port nos. 01 to 03 to the Video Conferencing equipment).
• The user will program the routing of IC calls to the Video Conferencing equipment using DDI routing table
for placing IC video conferencing calls.
• Video Conferencing generally requires 6 B-channels. But Video Conferencing can also be done at
lower bit rates also using the "aggregation" of 6 B channels which must be supported by the VC
equipment.
• Please note that a Video Conference call cannot be transferred or kept on Hold.
• The computers connected in the LAN can browse the net through the BRI.
• Remote LAN Access the Computers in the LAN can access the computer/computers in LAN at the
remote end (Branch office/Home office).
• For this, the ETERNITY will allocate data channels only on the BRI-TE port so as to leave other channels
for speech calls when the system detects the call to be a data call.
• Similarly, a Remote Computer can be accessed (Remote LAN Access) by dialing the Remote users'
number (The remote end PBX should be so programmed that the call made to a number lands directly on
the Router.) This establishes a permanent connection between the two Routers (and hence two LAN
networks).
• Now the user at PBX-A can access the computer in LAN at the remote end in the same way as accessing
another computer on the same network.
• However, while routing call on the BRI-NT port, the PBX will check that the data channels reserved for data
communication on the BRI-NT port are enough to establish the call. Otherwise the call will be rejected.
• The call will be rejected if the number of channels reserved for data calls, are already busy with one data-
call.
BRI Parameters
Hardware Slot-Port
Use following command to de-assign the hardware slot and the hardware port assigned to the BRI software port.
1106-BRI-00-00
Name
This parameter is used to enable/disable the port. When the Port configured in TE mode is disabled, it will not be
allotted to the user on grabbing the port. Instead the user will get error tone. Also no IC call will be allowed.
Likewise, when the port is configured in NT mode is disabled, the IC call will not be allowed to land on this port. The
port will be treated as absent and accordingly other activities will be performed like routing the call to other stations
in the group, etc.
Use following command to program Orientation Type for the BRI port:
6204-1-BRI-Orientation Type
6204-2-BRI-BRI-Orientation Type
6204-*-Orientation Type
Where,
1 Terminal
2 Network
3 Tie Line
By default Type = 1.
When Orientation=Terminal, the port will be regarded as trunk. All the trunk related parameters will be applied.
When Orientation=Network, the port will be regarded as station. All the station related parameters will be applied.
When Orientation = Tie-line, the port will be regarded as station for all incoming calls on it and as a trunk for all
outgoing calls made through it.
This parameter is relevant only when BRI port is used as Network Interface, i.e BRI NT mode.
Use the following command to select the Interface type as Point-to-Point or Point-to-Multipoint:
6226-1-BRI - Interface Type
6226-2-BRI-BRI-Interface Type
6226-*-Interface Type
Where,
Interface Type is
1 for Point-to-Point
2 for Point-to-Multipoint
Default =Point to Point
When ISDN Phones are to be connected with BRI NT of the ETERNITY, this parameter should be set as 'Point to
Multipoint'.
When an ISDN PBX is to be interfaced with BRI-NT of the ETERNITY, this parameter should be set as 'Point to
Point'.
Use following command to program the ISDN BRI Switch Variant of the BRI port:
6203-1-BRI-ISDN BRI Switch Variant
6203-2-BRI-BRI-ISDN BRI Switch Variant
6203-*-ISDN BRI Switch Variant
Where,
01 ATT_4ESS
02 ATT_5ESS
03 AUSTRALIA
04 AUTO CONFIG
05 DMS_100
06 ETSI_NET3
07 NTT_INS64
08 SWV_HONG_KONG
09 US_NI1
10 US_NI2
11 VN_X
OG Reference ID
IC Reference ID-WH
IC Reference ID-BH
IC Reference ID-NH
Use the following command to assign a Trunk feature Template to the BRI Port:
Cost Factor
Use the following command to assign a Station Basic Feature Template to a BRI port:
5509-1-BRI-Template Number
5509-2-BRI-BRI-Template Number.
5509-*-Template Number
Where,
BRI is from 01 to 32.
Template is the number of the Station Basic Feature Template, from 01 to 50.
Use the following command to assign a Station Advanced Feature Template to a BRI port:
5609-1-BRI-Template Number
5609-2-BRI-BRI-Template Number
5609-*-Template Number
Where,
BRI is from 01 to 32.
Template is the number of the Station Advanced Feature Template, from 01 to 50.
Default: Template 01 is assigned to all BRI ports.
Layer 1 Mode
• This parameter is applicable for BRI port, when orientation type is 'Terminal'.
• The Public ISDN provides different types of lines in different countries as mentioned below:
• On Demand
• Always ON
• In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle.
• When the network places incoming call, it activates the layer 1 and places the call.
• When the terminal makes out going call, the layer 1 gets activated automatically.
• In this type of line, 'layer 1 down' indicates fault condition and calls get failed.
1 Always ON
2 On Demand
Select the 'Layer 1 Mode' parameter, depending upon the type of line, terminated on BRI port of ETERNITY.
• When Layer 1 mode is set as 'Always ON', the ETERNITY uses this BRI to place call only if the layer 1
is 'up'. When layer 1 goes 'down', ETERNITY considers the line as un-healthy and will not use this BRI
as destination port. ETERNITY will place the call using the alternate port programmed in the same
routing group.
• The default Layer 1 Mode is 'Always ON'. Hence, if the interfaced line is of the type 'On Demand', the
calls will not get routed through BRI port, unless the 'Layer 1 Mode' is changed to 'On Demand'.
• When the port is un-healthy, the ETERNITY routes the call using other healthy port. However, this
depends upon the member selection method and other ports programmed in the OG Trunk Bundle
Group (OGTBG). Refer chapter “OG Trunk Bundle Group” for more details.
TEI Negotiation
SE can select the Automatic or Fixed TEI negotiation for each BRI port as required. If Fixed TEI negotiation is
selected, the value of fixed TEI negotiation is required to be programmed.
0 Automatic (Non-fixed)
1 Fixed
Use following command to program TEI Negotiation value when programmed as Fixed:
6239-1-BRI-TEI Value
6239-2-BRI -BRI-TEI Value
6239-*-TEI Value
Where,
BRI is from 00 to 32.
TEI Value is from 00 to 63
By default, TEI Value is 00.
The System Engineer should take care of the following points when using the above commands for TEI negotiation:
• When you change the TEI mode on any port, the BRI card will get reset.
• If you have selected 'Fixed' mode, program the value as per the value of port connected at remote end as
explained below:
• TEI Value programmed in the BRI (NT) should match with the TEI value programmed in the Terminal
equipment connected with it.
• TEI value of BRI (TE) port of ETERNITY should match with the TEI value expected by the NT
equipment at other end.
This timer is relevant while receiving the called party number information in overlap receiving mode. It is not
relevant for overlap sending mode.
Idle Code
Use following command to program the Idle Code for the BRI port:
6207-1-BRI-Idle Code
6207-2-BRI-BRI-Idle Code
6207-*-Idle Code
Where,
BRI is from 01 to 32.
Idle Code is from 000 to 255.
The binary equivalent of the programmed value (000 to 255) is sent on the channel to signify that the channel is
idle. (or Unused) This setting depends on the network. Most commonly applicable values are 7F and FF (Binary
equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Use the following command to program number of channels reserved for data transmission:
6235-1-BRI-Channel Count
6235-2-BRI-BRI-Channel Count
6235-*-Channel Count
Where,
BRI is from 01 to 32.
Channel Count is from 00 to 02.
By default, Number of Channels is 02.
Use following command to reserve the number of channels for OG calls on a BRI port.
6236-1-BRI-Channel Count
6236-2-BRI-BRI-Channel Count
6236-*-Channel Count
Where,
BRI is from 01 to 32.
Channel Count is from 0 to 2.
By default, Channel is reserved for OG Calls for BRI is 2. Both the channels are used for receiving OG calls.
Use following command to reserve the number of channels for IC calls on a BRI port.
6237-1-BRI-Channel Count
6237-2-BRI-BRI-Channel Count
6237-*-Channel Count
Where,
BRI is from 00 to 32.
Channel Count is from 0 to 2.
By default, Channel is reserved for IC Calls for BRI is 2. Both the channels are used for receiving IC calls.
Use following command to program a Caller TON for the BRI port:
6221-1-BRI-Caller TON
6221-2-BRI-BRI-Caller TON
6221-*-Caller TON
Where,
BRI is from 01 to 32.
1 Unknown
2 International Number
3 National Number
5 Subscriber Number
6 Abbreviated Number
7 Reserved Number
Use following command to program a Caller NPI for the BRI port:
6222-1-BRI-Caller NPI
6222-2-BRI-BRI-Caller NPI
6222-*-Caller NPI
Where,
BRI is from 01 to 32.
1 Unknown
2 ISDN Numbering
3 Date Numbering
4 Telex Numbering
5 National Numbering
6 Private
7 Reserved
Use following command to program a Called Party destination TON for the BRI port:
6223-1-BRI-Called Party TON
6223-2-BRI-BRI-Called Party TON
6223-*-Called Party TON
Where,
BRI is from 01 to 32.
1 Unknown
2 International Number
3 National Number
5 Subscriber Number
6 Abbreviated Number
7 Reserved Number
Use following command to program a Called Party NPI for the BRI port:
6224-1-BRI-Called Party NPI
6224-2-BRI-BRI-Called Party NPI
6224-*-Called Party NPI
Where,
BRI is from 01 to 32.
1 Unknown
2 ISDN Numbering
3 Date Numbering
4 Telex Numbering
5 National Numbering
6 Private
7 Reserved
Call Budget
Refer the topic “Call Budget on BRI Trunks” for command strings.
Call Back
Use the following commands to program Call Back on BRI Trunk ports. To know more about this feature, refer the
topic Call Back on Trunk Ports.
0 Disable
1 Enable
1 DID
5 Operator
1 CLI Number
2 Alternate Number
Pause Timer
This Timer is used to provide delay for the number dialing by the BRI port. Pause timer will be applicable when any
'P' digit is configured in the DTMF number string which is to be outdialed as DTMF digits on BRI port. One of the
applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage Dialing”.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY will out dial
the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
This parameter decides for how much time the DTMF digit will be ON, while out dialed by the ETERNITY. One of
the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage Dialing”.
This parameter decides how much time the pause (gap) should be present between two digits while dialed by the
ETERNITY. One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage
Dialing”.
Use the following command to program 'Category (Logical Partition)' for BRI port:
6213-1-BRI- Category
6213-2-BRI-BRI- Category
6213-*-Category
Where,
BRI is from 01 to 32
Category is from 1 to 3.
1 for CO
2 for Leased
3 for Private.
By default, Category is CO.
This command is used to enable Gateway Application-Answer Signaling on BRI trunk. Refer chapter “Gateway
Application-Answer Signaling” for more details.
Use following command to set flag for Gateway Application-Answer Signaling on BRI trunk:
6206-1-BRI-Gateway Application-Answer Signaling Flag
6206-2-BRI-BRI-Gateway Application-Answer Signaling Flag
Flag Meaning
0 Disable
1 Enable
Use following command to program DTMF digits string to be dialed as Gateway Application-Answer Signaling:
6212-1-BRI-Gateway Application-Answer Signaling DTMF String
6212-2-BRI-BRI-Gateway Application-Answer Signaling DTMF String
6212-*-Gateway Application-Answer Signaling DTMF String
Where,
BRI is from 01 to 32.
DTMF Digits allowed for DTMF string are from (0 - 9), *, #, A, B, C, D.
Maximum 4 DTMF digits can be programmed. If you need less than 4 digits for DTMF string, terminate the
command using #*.
A #4
B #5
C #6
D #7
* **
# ##
Debug Code
Debug information of various parts of the card can be obtained on the COM port of the card.
Use following command to get appropriate debug information for the BRI port:
6291-1-BRI-Level-Code
6291-2-BRI-BRI-Level-Code
6291-*-Level-Code
Where,
BRI is from 01 to 32.
Level is from 1 to 4 (As shown below).
Code is the value for the specified level to turn ON the debug for the parameters Code range = 000 to 255. Code
value '000' for each level will turn off that level's debug.
Level 1
Code Meaning
001 Primitives
002 State
004 Variables
016 LAP
032 NLS
Level 2
Code Meaning
004 Layer 4
Level 3
Code Meaning
Level 4
Code Meaning
002 OS Task
By Default Debug = off for all BRI ports for all levels.
• If '004' decimal value is entered as Debug Code for Level 1, then its binary equivalent '100' (0000100)
indicates that debug for "Variables" will get enabled.
• If "007" decimal value is entered as a Debug Code for the Level 1, then its binary equivalent "111"
(00000111) indicates that debug for "Primitives", "States" and "Variables" will get enabled.
Relevant Topics:
1. “ISDN-PRI” 1285
2. “Multi-Stage Dialing” 1331
3. “Gateway Application-Answer Signaling” 1236
4. “Logical Partition” 1313
5. “Call Budget on Trunk” 874
What’s this?
• This topic explains the connection of PRI line, and application of PRI.
• PRI port of the ETERNITY configured for NT mode can be connected to the PRI Port of another PBX
configured for TE mode. In such case, the ETERNITY will behave as Transit Exchange.
• Dialing method on the PRI port is not programmable. The PRI port (configured for TE mode) will send the
called party number in Enbloc mode.
• All the switch variants are applicable to the PRI port whether programmed in TE or NT mode.
Applications:
Applications for PRI-NT Port are as described below:
• Video conferencing system connected to the PRI-NT port with IC/OG calls.
• Data Calls support.
• Networking of PBXs.
PBX
ISDN PRI TE
Video
PRI NT Conferencing
System
• Connect the Video Conference system (H.320) to the PRI-NT port of the ETERNITY.
• This feature works only if it is supported by the Service Provider. Video Conference is established mainly
by the Video Conferencing (VC) equipment.
• Video Conferencing requires 6 B-channels. But Video Conferencing can also be done at lower bit rates
also using the “aggregation” of 6 B channels which must be supported by the VC equipment.
• The VC equipment uses two methods:H.221 and H.242 BONDING, for aggregation.
• More than one Videoconferencing system can be connected to the ETERNITY to each PRI-NT port.
• It also supports internal calling like, one Video Conferencing system to another.
• Program the OGTBG (OG Trunk Bundle Group) such that the user gets at least 6 B-channels of the same
PRI port.
• Go OFF-Hook.
• If VC at the called party responds, the call is established which occupies 6-B Channels.
• If 6 B-channels on the same PRI port are not available, user at VC will get busy tone.
• The user at calling party’s VC gets dial tone of the ISDN exchange.
• The user at VC starts dialing the destination number. The destination number is sent in keypad IE by the
VC user to the PRI-NT port whereas it is dialed on the PRI-TE port in the method programmed for the port
i.e. Enbloc.
• Rest of the signaling is done between the VC equipment and the called party’s VC equipment.
• If VC equipment supports Phone Book feature, user can make call using the feature.
• At the end of VC, all the 6 B-channels are freed and available for other users.
• It is preferable that the SE will assign an OGTBG to the Video Conferencing equipment in such a manner
that the same group is not assigned to any other station. This is to allow Video Conferencing call at any
time.
IC Video Call:
• Following steps are followed for IC call:
• Program the system to place IC video calls to the PRI-NT port using DDI Routing Table.
• For this, program the DDI Routing Table. Refer related chapter.
• You can also prepare the Trunk Landing Group (Routing Group) to land the call to PRI-NT port. For this
purpose, Routing Group is to be modified.
• For data-communication, connect the Router supporting ISDN-PRI or a Computer with ETERNITY ME
Card T1E1PRI to the PRI-NT port as shown below:
ISDN PRI TE
PRI NT Hub
Router
• The computers connected in the LAN can browse the net through the PRI.
• Remote LAN Access the Computers in the LAN can access the computer/computers in LAN at the remote
end (Branch office/Home office).
• For this, the ETERNITY will allocate data channels only on the PRI-TE port so as to leave other channels
for speech calls when the system detects the call to be a data call.
• Similarly, a Remote Computer can be accessed (Remote LAN Access) by dialing the Remote users’
number (The remote end PBX should be so programmed that the call made to a number lands directly on
the Router.) This establishes a permanent connection between the two Routers (and hence two LAN
networks).
• Now the user at PBX-A can access the computer in LAN at the remote end in the same way as accessing
another computer on the same network.
• However, while routing call on the PRI-NT port, the PBX will check that the data channels reserved for data
communication on the PRI-NT port are enough to establish the call. Otherwise the call will be rejected.
• The call will be rejected if the number of channels, reserved for data calls are already busy with one data-
call.
2001
PBX-A PBX-B 3001
ISDN PRI TE
ISDN PRI TE
PRI NT PRI TE
• A station 2001 of PBX-A can call a station 3001 of PBX-B by dialing 3001.
• Such a call can be routed using CUG table of PBX-A. Refer chapter “Closed User Group (CUG)”.
Method 1
• Program (for PBX-A) an OGTBG containing the PRI-NT port and assign Trunk access code to it.
• Grab the PRI-NT by dialing access code. User of 2001 will get dummy dial tone of the PBX-A.
Method 2
• User of 2001 can dial trunk access code of PBX-A, and dial a trunk access code of PBX-B. He will be
connected to ISDN network through PBX-B.
• Then he can make OG call on PRI-NT port as per method programmed for the PRI-NT port. (For example,
PRI-NT port can be programmed as a Trunk port and when station user of PBX-B grabs the PRI-NT port,
his call will be placed on the destination programmed for PRI-NT port in the Trunk Landing Group).
• Using CUG feature of PBX-B, 3001 can call 2001 through PRI-NT port. Refer chapter “Closed User Group
(CUG)” for more details.
• 3001 can also dial Trunk Access Code to grab its PRI-TE port. He will get dial tone of PRI-NT Port of PBX-
A. Now 3001 can dial 2001 or trunk access code to make a call to the ISDN network connected to PBX-A.
• Trunk software ports are automatically assigned to the PRI port by the system depending on the slot in
which they are inserted.
• Please take care to terminate the PRI line on the HDSL interface only of the ISDN modem.
• The DDI Routing and Routing Tables shall be programmed as explained in chapters on DDI.
Relevant Topics:
1. “Software Port and Hardware ID” 1462
2. “T1E1 Trunks” 1608
3. “DDI Routing Table” 1081
What’s this?
Extension users often have to dial access codes for specific functions like dialing a feature code, making an internal
call, making an external call, etc.
ETERNITY supports Macros, using which, you can abbreviate long number strings for regularly used functions in to
macros and assign them to a DSS key on a DKP/Extended IP Phone extension.
You can also assign Macros on SLTs that have special keys.
How it works
• Extension 2001, frequently sets Call Forward-All Calls to an external number 26550333.
• To do this, each time, Extension 2001 must dial 131-Trunk Access Code-26550333-#*.
• Instead of having to dial this lengthy number string, a Macro can be created for Call Forward-All Calls to
External number.
• If Extension 2001 is an Extended SIP Phone or a DKP, the Macro can be assigned to a DSS key on the
phone.
• Instead of dialing this number string, the user of Extension 2001 can simply press the DSS key on which
this Macro is assigned.
• Thus when the DSS key on which a Macro is assigned is pressed, the corresponding access code is
executed.
ETERNITY also supports Macros for SLT which have special keys. When each of these keys is pressed, a special
number string, which you can program is dialed.
For example, an SLT instrument has 5 special purpose keys. When these keys are pressed, the strings *50, *51,
*52, *53, *54 programmed on these keys are dialed out.
You can create Macros for the strings dialed out using the special keys, whereby the string dialed by each of these
special keys is associated with a particular function. For example, the special key for dialing *50 is associated with
Call Forward -All Calls to an external number. So, when the extension user presses *50, the system receives this
string and takes appropriate action, i.e. interprets it as call forward to the external number, and sets call forward.
Thus, each time the extension user presses the special key *50, the system considers that the extension user has
dialed 131-Trunk Access Code-26550333-#*.
• Click Macros.
The Macros page opens. Each macro is stored against an index number. By default the Macros Number
String and Access Code are blank when the system is operated in the Enterprise Mode.
• In the Number String field, enter the strings the system should consider as command when the DSS
Key on the DKP/Extended IP Phone.
For detailed instructions on assigning a macro to a DSS key of a DKP, see “DSS Keys Programming”.
For instructions on assigning a macro to a DSS key of the Extended IP Phone, see “Matrix Extended IP
Phone Settings” under Configuring SIP Extensions.
• In the Number String field, enter the strings the system should consider as command when the special
key on the SLT is pressed.
• In the Access Codes field, enter the strings sent by the SLT on pressing the special function key.
For example, if the SLT sends the string ‘*53’ to the ETERNITY when the function key for Alarms is
pressed, enter the string 163 (the feature access code for Voice-guided Alarms) in the Number String
field, and enter the string *53 in the corresponding Access code field.
The Access Code that you assign here in Macros must not conflict with any other Access Codes in the
Dial Phase. See “Access Codes”.
When ETERNITY is operated in Enterprise mode, the default Macros Number String and Access Codes
are blank. So, when you dial this command, the values in the Macros will become blank.
When ETERNITY is operated in the Hotel Mode, the Number Strings and Access Codes assigned by
default will be restored when you dial this command. See the topic Customer Profile in the ETERNITY
Hospitality System Manual to know more.
• Exit SE mode.
What’s this?
ETERNITY offers a facility—Last Caller Recall— to trace the extension that last made the call to your extension.
How it works
• When the called extension answers, speech is established between B and the called extension user.
How to use
For EON & Extended IP Phone Users
• Lift Handset
• Press DSS Key assigned to Last Caller Recall.
OR
• Go OFF-Hook
• Dial 1092
The system dials out the extension number that last called your extension.
What’s this?
The system redials the last external number string dialed from an extension.
How it works
• Extension A dials the feature access code for ‘Redial’.
• If Extension A is an SLT, the system dials the last external number dialed from Extension A using the same
trunk access code used for dialing that number.
• If Extension A is a DKP or an Extended IP Phone, all external numbers dialed by Extension A are
displayed on the phone’s LCD.
• Extension A may select the number to be dialed out. The system will dial out this number using the same
trunk access code used for dialing this number.
• If Extension A has ‘Dynamic Lock’ set and uses Redial feature, the system will check for Toll Control as
per the Lock Level set for Extension A before dialing out the number.
How to configure
No particular configuration is required for this feature to work. Redial is included in the Basic Features allowed to all
extensions by default in their “Class of Service (COS)”. So, all extensions can use the Redial feature.
How to use
For EON & Extended IP Phone Users
• Press Redial Key.
Or
• Dial 7
A List of external numbers last dialed will appear on your phone’s display.
• Scroll to select the desired number.
• Press Enter key.
• The system dials out the external number.
What’s this?
This type of Least Cost Routing is used in countries where the same service provider offers local call and long
distance calling services. These service providers allow subscribers to select the service provider or Carrier for long
distance calling.
For example,
Service
Provider B
Service
Provider D
A subscriber of Service Provider A must grab trunk lines of Service Provider A to call other subscribers in the local
area.
However, when the subscriber of Service Provider A wants to make a long distance call, the subscriber must dial a
prefix to select the a carrier (trunk) of the desired long distance, Service Provider B, C and D. Thus, the subscriber
accesses a secondary service provider by dialing a short code or prefix for long distance calling.
This feature works on the basis of “Automatic Number Translation”. Using Automatic Number Translation,
ETERNITY adds the code of the appropriate secondary Service Provider to the number string dialed by the
extension user to route the call to the desired secondary Service Provider.
How to configure
To use this feature, you must do the following:
• Configure the Automatic Number Translation Table. In the Automatic Number Translation Table, in the
Dialed Number String column, enter the long distance numbers that extension users will dial. In the
Substitute Number String column, enter the Prefixes of the Secondary Service Providers which should be
added to the dialed numbers. For example, the code ‘961’ for Service Provider B must be prefixed to the
number ‘2630555’ dialed by extension users, you must enter ‘2630555’ in the Dialed Number String
column and ‘9612630555’ in the Substitute Number String column.
• Create “OG Trunk Bundle Group” that includes the OG Trunk Bundle you created with Automatic Number
Translation as member.
What's this?
Certain features of ETERNITY require the purchase of a license. When you buy ETERNITY, you get a unique
license number for your product and a set of 'license-free' features are loaded in the system. Described below are
the features that require license.
PLCC
This functional module contains a set of special features supported by the ETERNITY when it is deployed in a
Power Line Carrier Communication Network of electric utilities. When you buy the license for this module, the
following features will be enabled:
• Express Signaling.
Hospitality
This functional module contains a set of special telephone and guest/patient management features for hospitality
and accommodation establishments like hotels and hospitals, which ETERNITY supports when it is deployed in a
hotel or hospital. When you buy the license for this module, the following features will be activated:
• Room Shift
• Check-In, Check-Out
• Floor Service
Dealer Board
You need to buy a Dealer Board license for the DKP ports to which you want to connect and use the digital key
phone, EON, as a Turret.
Gateway
When ETERNITY is used as a Universal Gateway, a license is required to activate the Gateway functionality.
SIP Extensions
ETERNITY supports up to 999 SIP Extensions, depending on your model. With a license, you can register SIP-
enabled devices with the VoIP Card of the ETERNITY. Without a license you cannot register any SIP Extension.
• Multi-Party Conference
• SMDR Buffer (to increase capacity from 200 each of Incoming, Outgoing and Internal Calls to 5000
Incoming, 6000 Outgoing and 1000 Internal Calls)
• SMDR Posting
• Multi-stage Dialing
• RCOC
• BCCH Selection
You may activate your License Online. For this, keep the following items ready:
• A valid, unique User ID and Password from the Matrix License Support Centre.
• Access to Internet.
• Open Jeeves.
You may view the features and functions that are currently available to you under Service Profile.
• Enter http://115.118.161.162:81/matrixlicense in the address bar of the new window and press the Enter
key on your keyboard or click Go on the address bar.
• Enter your User Name and Password provided by Matrix and click the Login button.
• In the Current License Key field, type the current product license key you noted from the License
Management page of Jeeves.
• Click Details. The details appear in the fields Product Family, Product Name, Product Variant.
• Click the Activate button and wait for a few seconds, as the activation is initiated.
On successful activation, the confirmation message will appear on your screen along with the activation
date and time.
You will also be sent a confirmation mail to your e-mail ID (registered with Matrix).
• Go back to the Jeeves window (or log in as System Engineer again, if your session has ended).
The Service Profile on this page will be updated according to the license.
If you are unable to use Online Activation of the License Key or have no internet access, contact the Matrix
License Support Centre for assistance in generating the new License key.
• Open Jeeves.
Under Service Profile, you may view the functional modules and features that are currently available to
you.
• Send your Current License Key and the License PIN (on the Voucher) to the Matrix License Support
Centre.
• In Enter License Key, enter the New License Key you obtained from Matrix.
The current License Key and Service Profile will remain unchanged when the system is set to default or
the firmware is upgraded.
What’s this?
Live Call Screening enables extension users to screen callers before attending their calls on their extensions.
How it works
To be able to use this feature,
• the extension users who are to be provided this feature must have a Personal Mailbox assigned, and have
this feature allowed to them in their Class of Service.
• the extension users who want to use this feature must set Call Forward to Voice Mail System on their
extension.
• the extension on which this feature is used must be a DKP or an Extended IP Phone.
With the above pre-requisites fulfilled, this is how Live Call Supervision will work:
• B calls A, and is transferred to A’s mailbox (as Call Forward to Voice Mail System is set).
• The VMS Auto Attendant offers B the option to leave a message in the mailbox of extension A.
• As B starts to record a message in A’s mailbox, the speaker of the DKP/Extended IP Phone of A gets
turned ON for the duration of the Live Call Screening Timer (configurable; default: 10 seconds).
• If A wants to answer the call, A can go Off-Hook. A gets connected to the caller and the system stops
recording the message in the mailbox.
• If A does not answer the call, the speaker is turned Off automatically on the expiry of the Live Call
Screening Timer.
• Also, after listening to some part of the message, if A finds that the call is not important, A can ignore the
caller by dialing any digit. When A dials any digit, the speaker of the DKP/Extended IP Phone is turned
OFF, but the message recording in the mailbox continues.
• Live Call Screening enabled in the “Class of Service (COS)” for the time zone in their “Station Basic
Feature Template”, as required.
If required, you may also change the duration of the Live Call Screening Timer. See “System Timers and Counts”
for instructions.
How to use
What’s this?
Using Live Call Supervision, any extension can know the last external number dialed by another extension, even
when that extension is in speech with an external party.
This feature is useful for supervisors who want to know where their subordinates are calling.
This feature is supported on DKP and Extended IP Phone extensions, and on SLT extension which have CLI
phone.
How it works
• A is the supervisor of B.
• When A requests Live Call Supervision for B’s extension, the system retrieves the last external number
dialed by B and presents it on the display of A’s phone.
• If the last number dialed by B is an internal number, A will get error tone, as the system supports live call
supervision of external calls only.
Live Call Supervision can be used also when the extension being supervised is in speech with an external
party.
How to configure
To be able to use Live Call Supervision, extension users must have this feature enabled in their “Class of Service
(COS)” in their “Station Basic Feature Template” for the required time zones.
How to use
For EON & Extended IP Phone Users
• Press DSS key assigned to Live Call Supervision
OR
• Dial 1098
• Enter the Extension number to be supervised
What's this?
Logical Partitioning is used to restrict the flow of call traffic between PSTN and Private Networks as well as
between PSTN and VoIP networks.
This feature may be used in countries where such restrictions are mandated by telecom regulations. For example,
in certain countries, calls from VoIP to Public Networks (PSTN, Public Land Mobile Network) are not allowed.
Thus, local telecom regulations may either disallow termination of lines from both networks on the same equipment
or may allow lines from both networks to be terminated on the same equipment, provided the equipment is
designed to restrict flow of call traffic from these networks. For example, the Telecom Regulatory Authority of India
allows termination of lines the PSTN and VoIP Networks in the same equipment, only if these lines are logically
partitioned. Termination of lines from both these networks in the same equipment without a logical partition
constitutes an offence.
How it works
Logical Partition is applied on the Trunk ports. Trunk ports are assigned to any of the following categories according
to their installation scenario:
• Category 1: Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network) are assigned this
category.
• Category 2: Leased lines terminated in the trunk ports are assigned this category.
• Category 3: Trunk ports used to interconnect two PBXs are assigned this category. For example, QSIG
used on T1E1 port or the TWT of ETERNITY is interfaced with FXS of other PBX for expanding
configuration.
• These are default Categories. You have the flexibility to define each category according to suit your
preference. For example, you may, if you so prefer, define Leased lines to Category 3 and Trunk ports
connecting two PBXs to Category 1.
• As calls from VoIP to Public network trunks are always be restricted, VoIP trunks cannot be assigned to
any of the three categories of the trunks as explained above and therefore do not require any
classification.
For each of the above categories, including VoIP trunks, the System Engineer can program the calling
permission, i.e. whether to 'allow' or 'restrict' the calls across and within these categories and with the VoIP
trunks.
By default, calls are restricted for all categories, except within VoIP, which includes both SIP trunks and
SIP extensions.
Depending on the calling permission programmed between the trunk categories, the system will allow or
deny the calls on the trunks.
• Now, assign the categories to the different Trunk Type, by programming the parameter 'Category (Logical
Partition)' in the Trunk Port Parameters of the respective trunk types: TWT, BRI, T1E1, E&M and Mobile
Port.
• Open the desired Trunk Port Parameters page by clicking the link. For example, to assign a Category to
TWT ports, open the TWT Hardware Template page.
• Scroll with the horizontal bar to reach the column 'Category (Logical Partition).
• Now, define the call permission for each category, i.e. 'allow' or 'restrict' calls. For example, allow/restrict
calls from Category 1 to Category 1, from Category 1 to Category 2, from Category 1 to Category 3 and so
forth.
• First, program the parameter 'Category (Logical Partition)' in the Trunk Port Parameters of the respective
trunk types. For SE Commands, refer the topics:
Flag is
0 for restrict calls
1 for allow calls
Default: 0 (i.e. restrict calls) for Category 1 to 3 and 1 (allow calls) for VoIP.
• Exit SE mode.
• By default for countries other than UK and USA calls within and between all categories of trunks are
restricted. In other words, trunk to trunk calls are not allowed.
• In UK and USA the calls within all categories of trunks, including VoIP are allowed.
• When call permission is restricted between two categories of trunks and/or between any category of
trunk to VoIP, following feature interactions will apply:
• Call Transfer: Trunk to Trunk Transfer between restricted categories of trunk will not be allowed. If
the user attempts trunk-to-trunk transfer between restricted trunks, Error Tone will be played.
• Raid: If a user using DISA attempts to Raid a conversation of an extension with a trunk to which call
permission is restricted, the Raid attempt will fail and the user will get an Error Tone.
• Conference: An extension user will not be able to include restricted trunks in a 3 party or multi-
party conference. An Error Tone will be played when s/he attempts it.
• Dial-In Conference: Participation in a Dial-In Conference from trunks with restricted call permission
is not allowed.
• External Call Forward: In the case of DID, DISA or when transferring a trunk call to an extension,
if the extension has set call forward to an external number, the system will allow the call only if the
call permission between the source and destination trunk is allowed. Otherwise, an Error Tone will
be played to the user.
• Hotline: When a user has logged into DISA and the extension being used for the DISA login has
the Hotline - Trunk or Hot outward dialing (HOD) feature enabled, the system will allow the call
between the source trunk (from where the DISA login is made) and the destination trunk (which is
used as Hotline Trunk) only if calling is permitted between them. Otherwise an Error Tone will be
played to the DISA caller on the expiry of the Hotline Timer.
What’s this?
Using Meet Me Paging, extension users can announce the name of the person they want to talk to on the Page
zone. The called person can respond to the announcement, by dialing, from any extension, the Meet Me Paging
code and the extension number of the caller.
When making a Meet Me Paging call, extension users can announce the extension number from where they are
calling, so that the called person knows which extension number to dial.
This feature is of great use to Operators. Using this feature, the Operator can put an important call on hold and
track the extension user who is not at the desk using Meet Me Paging. Once the person is located and calls the
Operator, the Operator can transfer the call to the person at the current location.
How it works
• The Operator receives a call for B.
• Operator calls B’s extension, but B is not at the desk.
• Operator uses Paging and makes the announcement for B, asking B to call the Operator.
• B is near extension 201. B dials Meet Me Paging Code and dials the extension number of the Operator.
• B is in speech with the Operator.
How to configure
To be able to use Meet Me Paging, extension users must have the feature“Paging” in their Class of Service.
How to use
If you are the Called Party (for whom the message has been announced):
• Lift the handset.
• Press the DSS key assigned to Meet Me Paging.
OR
• Dial 1093
• Dial the number of the paging extension.
• Speak with the paging extension user.
If you are the Called Party (for whom the message has been announced):
What’s this?
The Message Wait feature of ETERNITY enables extension users/Operator to set Message Wait on other
extensions to deliver important messages.
If the extension user has a mailbox assigned, the Message Wait feature indicates to the extension user, the arrival
of new messages in the user’s mailbox.
Thus, Message Wait can be set by extension users as well as by the Voice Mail System.
How it works
• The Operator has an important message to communicate. So, the Operator sets Message Wait on
Extension A, using the Message Wait key (if configured) or by dialing the feature access code.
• Extension B tries to reach Extension A, and sets Message Wait on Extension A, using the Message Wait
key (if configured) or by dialing the feature access code.
• Message Wait will be indicated to Extension A according to the Type of Message Wait Notification set for
Extension A. This may be in the form of a Stuttered Dial Tone, a Voice Message, Ring, or LED Lamp.
• If Extension A is a DKP or an Extended IP Phone and has DSS key assigned for Retrieve New Message,
the LED of this key will glow to indicate new message wait.
• Now, Extension A can dial the feature access code to retrieve Message Wait, or press the Retrieve
Message Wait Key, if assigned.
• The system will call the extension that first set Message Wait on Extension A. In this case, the Operator. If
the Operator is busy, the system will place the call on Extension B. The system will try to call the
extensions that set Message Wait until the call is answered.
• The extension that set Message Wait on A gets the CLI of A as Message Wait. A can now deliver the
message.
• The LED of the Retrieve Message Wait key, if assigned, on Extension A will be turned off after all message
wait set by other extensions on Extension A have been served.
• There is a new message in A’s Mailbox. The VMS indicates this to Extension A as per the Type of
Message Wait Notification set for Extension A. This may be in the form of a Stuttered Dial Tone, a Voice
Message, Ring or LED Lamp.
• If Extension A is a DKP or an Extended IP Phone, the Voice Mail key on the phone will also glow to
indicate the arrival of a new message.
• If the Retrieve Message Wait key is assigned on the DKP/Extended IP Phone of Extension A, the LED of
this key will also glow simultaneously to indicate arrival of the new voice mail.
The VMS answers the call. After Extension A has listened to the new messages, the LED of the Voice Mail
key is turned off.
The LED of the Retrieve Message Wait key, if assigned, will also be turned off.
Voice Mail has priority over extension Message Wait set by extensions. If an extension has both
Message Wait and new Voice mail, and when the extension presses the Retrieve Message Wait key or
dials the feature access code to Retrieve Message Wait, the call will be placed first to the Voice Mail
System. When the extension user presses the Retrieve Message Wait key again, the call will be placed to
the extension that set Message Wait.
• If you want voice message to be played as message wait notification, record and assign a Voice Module.
Refer “Voice Message Applications” for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait
Notification will not be played if there are already 4 being played simultaneously. In which case, Stuttered
Dial Tone will be played for Message Wait Notification, when the extension user goes OFF-Hook.
Ring
• When a new Message Wait is set on the extension, the system will play Message Wait Ring (Short, Fast)
on the extension. See “Distinctive Rings”.
• The extension will ring for the duration of the Message Wait Ring Timer (configurable; default: 30
seconds). If the call is not answered within this timer, the system will ring on the extension again for as
many times as the Message Wait Ring Count (configurable; default: 10 times), and at the interval set as
the Message Wait Ring Timer Interval (configurable; default: 30 minutes).
• When the extension user answers the call, the user gets connected to the VMS or the extension that set
Message Wait.
Particulars Value
ON Time 100ms
Frequency 4Hz
DC Offset 48V
How to configure
To provide this feature to extensions, you must do the following configuration on the extensions:
• Enable the Message Wait feature in the “Class of Service (COS)” of the “Station Basic Feature Template”
of the extensions. This allows the extensions to set and cancel Message Wait on other extensions. Only
those extensions that have this feature in their COS can set or cancel Message Wait on other extensions.
By default, this feature is enabled in the COS of all extension types for all the time zones.
• Select the desired Message Wait Notification Type in the “Station Advanced Feature Template” of the
extensions.
• If you selected Voice Message as Message Wait Notification Type for an extension, you must also record
the desired Voice Message in a Voice Module and assign it to the Message Wait application. See “Voice
Message Applications” for instructions.
• If you selected Ring as Voice mail/Message Wait Notification Type for an extension, you may configure
the following Ring Parameters:
• Message Wait Ring Timer (default: 30 seconds)
• Message Wait Ring Count (default: 10 attempts)
• You may also configure the features ‘Message Wait’ and ‘Retrieve Message Wait’ on DKP and Extended
IP Phone extensions.
For instructions on assigning these features to DSS keys of a DKP, see “DSS Keys Programming”.
For instructions on assigning these features to DSS keys of the Extended IP Phone, see “Matrix Extended
IP Phone Settings” under Configuring SIP Extensions.
How to use
• Press DSS Key assigned to 'Retrieve Message Wait' or the Voice mail Key when the LED glows.
OR
• Dial 1077
For SLT
What's this?
ETERNITY offers mobility to its extension users whose nature of work keeps them from their desks frequently and
for longer durations.
Using mobility extensions, the extension users of ETERNITY can make and receive their calls from their current
(remote) location, placing calls through the system, and can access the system just as any other normal extension
of the ETERNITY.
How it works
ETERNITY supports two types of users:
• Station Users: they are extension users of ETERNITY to whom a dedicated physical station is assigned
on their desk.
• Virtual Users: they are extension users of ETERNITY who share a physical station, or may not have any
physical station allotted to them.
The facility of Mobility Extension is provided to both virtual and normal extension users using the features
“Direct Inward System Access (DISA)” and “Call Forward”.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How to configure
To provide Mobility Extension to users, the follow the steps described below.
• List out the Station Users and Virtual Users and configure them first. The software ports of SLT, DKP and
ISDN Terminals which are not assigned a physical slot - port, can be used as Virtual stations.
• Make sure that stations which are to be provided Mobility Extensions have the features Call Forward and
DISA enabled in their “Class of Service (COS)”185. Class of Service is to be programmed in the “Station
Basic Feature Template” assigned to the Mobility Extensions.
• Make a list of External numbers to which the Mobility Station users will forward their calls. Program these
numbers in the 'Allowed List' of Local, Regional, National and International Numbers, as appropriate.
• The Toll Control assigned to the station will be applied when a call is forwarded to an external number.
Make sure that the stations which are to be provided Mobility Extension have the required “Toll Control”
level for Call Forward to the External Numbers (the numbers you have programmed in the Allowed List).
Toll Control level is to be programmed in the “Station Basic Feature Template” assigned to the Mobility
Extensions.
185. It is possible to map the virtual user's flexible number (extension number) to a physical station. With this mapping, whenever a vir-
tual user's number is dialed, the call is placed to the assigned physical station. The physical station can be assigned by defining it
as the 'Landing Destination for Virtual Users' in the Station Advanced Feature Template assigned to the virtual user.
• Program the parameter 'DISA' in the “Trunk Feature Template” of the trunk lines which Mobility Extensions
users are to be provided access to. Make sure you select the option 'CLI Auth. Multiple Calls' in the 'DISA'
parameter of the 'Trunk Feature Template'.
• Make a list of numbers which the Mobility Extension users will use to access the ETERNITY from DISA
mode. Program this list of numbers in the "DISA-CLI Authentication Table".
Program this list of numbers in the 'Calling Number' field of the Authentication Table. Program the 'Port
Type' and 'Port Number' of the Station assigned to Mobility Extension Users in the 'Auto Login As' field for
the respective 'Calling Number' field. Refer the topic “Direct Inward System Access (DISA)” to know more.
How to use
The Mobility Extension Users of ETERNITY can use the features of ETERNITY from a remote location as
described below.
Receiving calls
To receive calls, the Mobility Extension User must set Call Forward on his station with an external number (mobile
number, landline number, etc.) as the destination number.
To make calls ring on the station and the external number simultaneously, the Mobility Extension User must
activate the Call Forward-Dual Ring feature on his station.
The Mobility Extension User can also choose where he wants to receive the calls during a particular time of the day.
For example, he can receive calls during a particular time of the day, i.e. Time Zone on his external number and
have his calls received by his Voice mail or the Operator or any other number during another Time Zone. To do
this, he must set “Call Forward-Scheduled" on his station. Dual Ring can also be set for Call Forward-Scheduled.
Making calls
The Mobility Extension User should make a call on the DISA enabled trunk of ETERNITY from the external number
and the system will provide the dial tone to the user after authenticating the external number with the help of the
DISA-CLI Authentication table.
On getting the dial tone, the Mobility Extension User can make internal as well as external calls as per the “Toll
Control” and “Class of Service (COS)” assigned to his Station.
The Mobility Extension User can also dial codes of the Personal directory and Global directory numbers to use the
feature Abbreviated Dialing.
Accessing Features
The Mobility Extension User can access the system features by dialing specific codes after making calls on the
DISA enabled trunk of ETERNITY, or after answering the calls received on his external number.
On-Hook #0
Off-Hook #1
Flash #2
Pause #3
Described below are instructions for Mobility Extension users on using different call management features.
Call Hold
To put a call on hold,
Call Transfer
To conduct a screened Call Transfer,
OR
OR
OR
Call Pick Up
To pick up the call of same Call Pick Up-Group,
3-Party Conference
To conduct a 3-Party Conference,
Multiparty Conference
To create a Multiparty Conference,
Call Forward-Busy
To set Call Forward-Busy,
Call Forward-Scheduled
To set/cancel Call Forward-Scheduled,
• Mobility Extension users can have Call Forward and Call Forward-Scheduled set on their extension by
the Operator or by another extension user.
• Using “Call Forward-Remote” and by setting “Call Forward-Scheduled” from the SA mode (using SA
commands or Jeeves), the extension user/Operator can set Call Forward and Call Forward-Scheduled
for any Mobility extension user. Refer the respective topics to know more.
What's this?
• There are some applications where we have to dial a fixed number string before dialing the actual number.
For example, using the ITC card (This card is provided by PSTN as prepaid card for calling the number
from BSNL line). In such application, the number is required to be out dialed after the call gets matured
and introducing some fixed number string (Calling card number) before dialing actual number. For
example, for making an ISD call using ITC card, you may be required to dial following sequence of digits:
• Dial first the number for using ITC calling card, for example, 1602233 (7 digits).
• Then, after the call is matured dial the 16 digit PIN number printed on the ITC card. For example: xxx..
(16 digits).
• After dialing the PIN number, dial the number to which you want to call i.e ISD number. For example,
yyy..(14 digits)
• Thus, you will have to dial 1602233 and 16 digit PIN number-i.e total 23 fixed digits every time, before
dialing the ISD number of 14 digits.
• To avoid pressing of such long digits for making call and to dial some digits with required delay, the feature
called 'Multi-Stage Dialing' is supported in the ETERNITY. Using this feature the ETERNITY will take care
for dialing the digits after the call is matured and dialing of digits with suitable pause time, before dialing out
the destination number.
• This feature works by suitable programming of following digits in the Number Lists for ANT feature: 0-9, #,
*, A, B, C, D, F, P, W, . (dot), +. Where A, B, C, D are the DTMF digits, F is for Flash, P is for Pause and W
is for Wait for Answer (period for the call is matured). Enter the special digits using specific Code for the
digit.
• While ETERNITY is dialing out the digits, user will get Call Proceeding Tone as configured in the system.
Refer System Parameters.
• Following parameters will be required, to configure this feature for the trunk port:
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic “License Management” to know more.
How to configure
Step 1
Program Dialed Number String using command '4751'. Refer chapters “Automatic Number Translation”.
• It is recommended to SE, to not to program Pause character "P" before "W" character for the number
string to be out dialed from Mobile port. Else, the GSM Module may get restart while out dialing the DTMF
digits without call maturity signal.
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
. (dot) #9
# ##
* **
W *1
Step 3
Enable 'ANT' for the trunk port using command 6702. Refer chapter “Outgoing Trunk Bundle” for more details.
Step 4
Assign the ANT Table to the trunk port using command 6702. Refer chapter “Outgoing Trunk Bundle” for more
details.
Step 5
Refer chapter “TWT Hardware Template” for command '5902' to program Pause Timer on TWT Trunk.
Refer chapter “T1E1 Trunks” for command '6109' to program Pause Timer on T1E1 Trunk.
Refer chapter “ISDN-BRI” for command '6209' to program Pause Timer on BRI port.
Refer chapter “Configuring Mobile Trunks” for command '8014' to program Pause Timer on Mobile port.
Refer chapter “Configuring SIP Trunks” for command '7720' to program Pause Timer on SIP trunk.
Refer chapter “E&M Feature Template” for command '6002' to program feature Pause Timer on E&M trunk.
Step 6
Refer chapter “TWT Hardware Template” for command '5902' to program DTMF ON Time on TWT Trunk
Refer chapter “T1E1 Trunks” for command '6117' to program DTMF ON Time on T1E1 Trunk.
Refer chapter “ISDN-BRI” for command '6210' to program DTMF ON Time on BRI port.
Refer chapter “Configuring Mobile Trunks” for command '8015' to program DTMF ON Time on Mobile port.
Refer chapter “Configuring SIP Trunks” for command '7725' to program DTMF ON Time on SIP trunk.
Step 8
Refer System Parameters for command '5311' for programming of Call Proceeding Tone.
Example:
Refer the figure given below:
Gateway 1 Gateway in
UK (0044)
• Scenerio1 (through Gateway): Route the Number starting with 0044 through Gateway 1 i.e.9898906335
through the TWT trunk port 001 after the call is matured (after period for Wait for Answer).
• Scenerio2 (with Calling Card): Dial the Number starting with '2' using the calling card; by dialing first
1602233 and then dialing the PIN number 1132121234#1P of the card after the call is matured (after
period for Wait for Answer).
Step 1
Program dialed number string '0044' in ANT Table-1
4751-1-1-01-0044-#*
Step 2
Program substitute number string '9898906335WP0044' in ANT Table-1
4751-1-1-01-9898906335 *1 #3 0044-#*
Step 3
Assign ANT Table-1 to Mobile port-001
6702-1-64-6-1
Step 4
Keep other parameters Pause Timer, Inter Digit Pause Timer, DTMF ON Time same as default value.
ANT Table-1:
1 0044 9898906335WP0044
2 2 1602233WP1132121234#1P2
32
Scenerio1:
• Program '0044' as Dialed String and program '9898906335WP0044' as Substitute string. Now when you
dial '00441159253724', the system will dial out the number as explained below:
• The system will dial out first, the Gateway1 number 9898906335.
• After the call is matured and delay for the pause timer, the system will dial the destination number:
'00441159253724'.
Scenerio2:
• Similarly, the system will process the call when you dial the number starting with prefix '2', for example,
2630555, using the calling card:
Relevant Topics:
1. “Automatic Number Translation” 825
2. “Configuring Mobile Trunks” 595
3. “Outgoing Trunk Bundle” 1348
4. “T1E1 Trunks” 1608
5. “ISDN-BRI” 1263
6. “Configuring SIP Trunks” 635
7. “E&M Feature Template” 565
8. “TWT Hardware Template” 530
What’s this?
The music played to extension users and external callers who are put on hold is called Music on Hold (MoH).
How it works
ETERNITY supports Music on Hold from an Internal Music Source as well as an External Music Source.
When a caller is put on hold, ETERNITY plays the music recorded in Voice Module 1.
You can play a voice message of your choice instead of music to the callers. The message may contain any
promotional information about your company or services provided by your organization, etc. For this, you must
record a Voice Module with the custom message and assign the Voice Module to the Music on Hold application.
If the option Routing Group is selected as the Alarm Notification Type for an extension, when the
extension goes Off-hook to answer an alarm call, and the extensions in the Routing Group for Alarm
Notification are busy, Music-on-Hold will also be played to the extension answering the alarm call.
Any external device meeting the specifications of the AIP can be connected with the system. The volume must be
set to an appropriate level such that the volume of the music on the trunks is neither too low nor too high. The
volume of the signal coming from this device must never increase beyond the specified limits.
For instructions on connecting an external music source to the AIP, see the topics “Installing ETERNITY ME”,
“Installing ETERNITY GE”, “Installing ETERNITY PE” as applicable to your model.
Code Meaning
1 Voice Module 01
By default, code is 1.
Use following command to select the type of music to be played when stations are kept on hold:
3552-Code
Where,
Code Meaning
1 Voice Module 01
By default, code is 1.
Use following command to select the type of music to be played when trunks are kept on hold:
3553-Code
Where,
Code Meaning
1 Voice Module 01
By default, code is 1.
• If all the extensions of the Routing Group you selected for Alarm Notification type are busy, the
extension user will be played MoH (MoH can be Voice Module 01 or through AIP).
Example:
Program the system such that when stations are kept on hold, they get Voice Module 01 on Hold and when trunks
are kept on hold, the caller gets music from AIP.
3552-2
3553-2
Relevant Topics:
“Background Music (BGM)”
“Voice Message Applications”
What’s this?
A ‘Number List’ is a group of number strings. ETERNITY uses Number Lists to support different features as Toll
Control, Call Duration Control, Call Taping, Call Back on Trunk Ports, Station Message Detail Recording (SMDR).
ETERNITY supports 16 Number Lists. Each Number List can contain upto 999 number strings. Each number string
consist of a maximum of 16 characters.
The number strings are stored against Location Index numbers in the Number List. The Location Index numbers
start from 001 to 999.
The default values of the Number Lists in the system are shown below:
001 00 0 00 * * * * * * * * * *
002 0 * * # # # # # # # # # #
003 1 # # F F F F F F F F F F
004 2 F F
005 3
006 4
007 5
008 6
009 7
010 8
011 9
012 *
013 #
014 F
015 +
016
: : : : : : : : : : : : : : : : :
999
By default, Number List 15 is assigned to Call Back Incoming Number List, and Number List 16 is assigned to Call
Back Outgoing Number List.
This requires you to program and assign two Number Lists: the Apply CDC to Number List and the Do Not Apply
CDC to Number List.
By default, Number List 07 is assigned to Apply CDC Number List, and Number List 08 is assigned to Do Not Apply
CDC Number List.
Call Taping
Call Taping allows you to record conversations of incoming and outgoing internal and external calls. When you set
Call Taping on an extension for external calls, the ETERNITY uses two Lists: Number List - Incoming Calls: this list
has the list of numbers of external callers whose conversation is to be recorded.
Number List - Outgoing Calls: this list has phone numbers of external called parties whose conversation is to be
recorded. The system matches the incoming and outgoing numbers with the respective lists to apply Call Taping.
By default, Number List 09 is assigned to Incoming Calls Number List, and Number List 10 is assigned to Outgoing
Calls Number List.
Toll Control
The Toll Control Call Privilege Type “Limited Calls” allows and restricts dialing of telephone numbers starting with a
particular digit or a particular area code or certain telephone numbers only. To apply Call Privilege type ‘Limited
Calls’ you must program an Allowed Number List with numbers that are to be allowed, and a ‘Denied List’ with
numbers that are to be restricted.
Similarly, SMDR Reports of Incoming and Outgoing Calls can be generated and printed for selected numbers by
programming and assigning a Number List.
By default, Number List 02 is assigned to Destination Wise storage of SMDR and SMDR Report Printing.
Refer the topics “Station Message Detail Recording”, “Station Message Detail Recording-Storage”, “Station
Message Detail Recording-Report”.
How to configure
Take a pen and a paper. Decide which of the above-mentioned seven features are to be used. Number List
according to the feature for which it is to be used.
Number Lists can be programmed from Jeeves or by dialing SE commands from a telephone.
• Enter each number string against a Location Index (refer to the list you prepared).
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
• Exit SE mode.
What’s this?
This feature helps the extension user to disconnect the speech transmission path in the middle of a conversation.
The extension user can still listen to the opposite party because the receiving path remains connected. Mute is
useful when you want to consult someone in the middle of a conversation, but do not want the opposite party to
listen to your discussion. You can Mute a call before making a call or during speech.
How it works
• A is in speech with B.
• A wants to consult to C in the room, but does not want B to hear their conversation.
• A presses the Mute Key.
• The transmit speech path from A to B is disconnected. The receive path remains connected.
• So, A will be able to hear B, but B will not be able to hear the conversation between A and C.
• When A has finished consulting C, to resume speech with B, A presses the Mute key again.
• The transmit speech path from A to B is restored. A and B are in speech again.
How to use
What's this?
When the handset of an extension is not placed correctly, it will not be possible for the Operator or any other
extension to call the extension. Also, incoming calls will not reach the extension, Alarms and Reminders cannot be
placed on that extension.
To avoid this inconvenience, the ETERNITY supports the feature 'OFF-Hook Alert', whereby the system detects
and informs the Operator of the extension phone that remains OFF-Hook accidentally.
How it works
To give the Operator an OFF-Hook Alert,
• When the Operator answers the call, s/he is played a confirmation tone, the text message remains on the
display.
• The Operator can inform the extension user to place the handset of the phone correctly.
• If the extension phone is an SLT, OFF-Hook Alert will be given to the Operator phone only. (The Operator
phone must be EON).
• If the extension phone is EON, OFF-Hook Alert will be give to both, the Operator phone and the extension
phone (EON).
• If the extension phone that is OFF-Hook is EON, the ETERNITY will activate 'OFF-Hook Alert' on the
extension phone, by playing the Error Tone continuously, on speaker to draw the attention of the extension
user.
• While the Error Tone for OFF-Hook Alert is being played on the extension phone, if the user presses the
Speaker Key, the Error Tone will continue to be played on the handset until it is replaced correctly.
How to configure
For this feature to work,
• the 'OFF-Hook Alert to Operator' flag must be enabled by the System Engineer in the 'System General
Parameters'.
• the Operator phone must be EON, the extension phones may be EON or SLT.
• Go to the parameter 'OFF-Hook Alert to Operator', select the check box to enable the flag.
What’s this?
The OG Reference Table is a set of general features that define the logic of building the DDI Number for a station
placing a call and sending it to the network. An OG Reference Table is assigned to SIP/T1E1PRI/BRI ports. This
table in conjunction with DDI Routing Reference Table builds the DDI Number. The ETERNITY offers 64 such
Tables each of which can be programmed as per the requirement.
How to use
• OG Reference Number-This is the reference number acts as an identifier to the translation logic
programmed in the OG Reference Table. Any number of table can have the same reference number. An
OG Reference ID can be assigned to ISDN, SIP, T1E1PRI and BRI trunks. For more details on the
complete translation logic please refer the topics “Direct Dialing-In (DDI)”, “DDI Routing Table”.
• Start Channel Number-This is the first channel number for the trunk to which the logic is applicable.
• Channel Count-The Total number of channels from the Start Channel Number of the port (Trunk) to which
the OG Reference Table is applicable.
• ISDN Number-Each ISDN Trunk is given an Installation Number by the Service Provider. This is the
combination of Main Number (MSN Number) and the first DDI Number. The Number is of maximum 16
digits. This is also known as ISDN Installation Number or just ISDN Number. The MSN number is given by
the service provider whereas the Directory Numbers can be selected by the user. However the number of
digits to be used for the Directory Number should be informed to the service provider.
• DDI Routing Reference ID-The ‘DDI Routing Reference ID’ programmed in the OG reference table,
provides mapping with the ‘DDI Routing Reference ID’ programmed in the DDI Routing Table. Using the
mapped entry of DDI Routing Table, the DDI number gets created from the flexible number. This DDI
number is sent as ‘Calling Party Number’ while making OG call.
How to configure
The commands explained below should be referred as:
To program a single port: XXXX-1
To program a range of ports: XXXX-2
To program all the ports: XXXX-*
Parameter No. 1 2 3 4 5
01 00 01 00 Blank 000
02 00 01 00 Blank 000
03-63 Same as 02
64 00 01 00 Blank 000
Parameter Value:
01-30
Relevant Topics:
1. “Direct Dialing-In (DDI)”
2. “ISDN-BRI”
3. “T1E1 Trunks”
4. “DDI Routing Table”
What’s this?
The Outgoing Trunk (OG) Trunk Bundle is set of parameters that completely define the grouping of similar
channels. The word channel refers to a speech path. By this definition, a channel can be a TWT trunk, an E&M
trunk, a Mobile port. Each speech path of the T1/E1 line is a channel. Bundles of similar trunks are formed. These
Bundles are used to form an OGTBG. The ETERNITY supports 128 OG Trunk Bundles.
How to use
The OG trunk bundle contains parameters like:
• Trunk Type: Specific trunk bundle consists of some of these ports. The different types of ports are TWT,
BRI, T1E1PRI, E&M, Mobile, and SIP.
• Trunk Number: Once the type of port is identified in the bundle, it is required to identify the number of that
port, because there can be more than one port for the given type.
• Trunk Count: This is the number of trunks to be kept in the same bundle. For TWT and E&M this could be
128. This value for the BRI and T1E1 Trunks is counted from the start channel. For example, if the port
type is TWT, port number is 002 and channel count is 025, then TWT channels 002 to 027 would be
grouped together. Consider a second example where channels 15 to 25 of T1E1PRI port are to be
programmed in one channel group, then the port type will be T1E1PRI, port number would be 5, start
channel number will be 15 and channel count will be 11.
• Rotation Type: This parameter shows which channel should be selected when the next call lands on that
port. For example, if ascending order is selected, the system checks 001-128, for first free channel and if
descending order is selected the system checks from 128-001.
• Ascending Order:
• Descending Order:
• Cyclic:
• Always the next channel is picked for a new OG call.
• ANT Table No.: This is a Table number in which Dialed Number Strings and corresponding Substitute
number strings are programmed at specific Index. This table number is assigned to the specific trunk port
from which the number is to be dialed. Refer chapter “Automatic Number Translation” for more details.
How to configure
Use following command to program the feature in an OG Trunk Bundle Number:
6702-1-OG Trunk Bundle Number-Feature Number-Code
6702-2-OG Trunk Bundle Number-OG Trunk Bundle Number-Feature Number-Code
6702-*-Feature Number-Code
Where,
OG Trunk Bundle Number is from 001 to 128.
Feature Number is from 1 to 6.
Feature Number
1 2 3 4 5 6
Parameter Values:
2 04 001-032 Descending
3 05 001-008 Cyclic
4 06 001-128
5 25 001-064
6 26 001-032
Relevant Topics:
“Automated Control Applications”
“Outgoing Trunk Bundle”
What’s this?
• OG Trunk Bundle Group provides efficient allocation of trunks to different stations.
• All the trunks connected to the system can be bunched in different groups called OG Trunk Bundle Group.
Maximum 8 Trunk Bundles can be put in one OG Trunk Bundle Group and 32 such OG Trunk Bundle
Groups can be formed.
• A station can be allotted different OG Trunk Bundle Group during different timings of the day.
How it works
System uses two methods while selecting a trunk from the OG Trunk Bundle Group: ‘Remember last trunk’ and
‘Don’t Remember last trunk’.
In ‘Remember last trunk’ method, the system remembers the last trunk used and allots next trunk in the group to
the station.
In ‘Don’t remember last trunk’ method, the system searches for a first free trunk from the group.
Start
Which
LCR type is
selected ?
Time based LCR Mixed LCR No LCR SP-SP LCR Number based LCR
End D
Is the
cheapest Yes
trunk free
? No
System allots the Is cheapest
No trunk to the station trunk free?
Are other No
End Yes trunks available
Are other in this group?
No System dials out
trunk available System gives busy
the number Yes
in this group tone to the station
System gives busy Select next cheap
Yes End
tone to the station trunk in this group
System waits for next
Select next action from user
cheapest trunk System waits for next
in this group action from the user
End
End Are
Yes other trunks
available with OG
trunk bundle
group ?
Select next cheapest
No
trunk within the group
End
How to configure
The commands explained below should be referred as:
To program a single port: XXXX-1
To program a range of ports: XXXX-2
To program all the ports: XXXX-*
Step 1
Use following command to make default OG Trunk Bundle Group:
1401-1-OGTBG Number
1401-2-OGTBG Number-OGTBG Number
1401-*
Where,
OGTBG Number is from 01 to 32.
Step 2
Use following command to set OG Trunk Bundle:
1402-1-OGTBG Number-Destination Index-OG Trunk Bundle
1402-2-OGTBG Number-OGTBG Number-Destination Index-OG Trunk Bundle
1402-*-Destination Index-OG Trunk Bundle
Where,
OGTBG Number is from 01 to 32.
Destination Index is from 1 to 8.
OG Trunk Bundle is from 001 to 128.
Step 3
Use following command to set Rotation Flag. For explanation on how to use, refer important point at the end of
chapter:
1403-1-OGTBG Number-Flag
1403-2-OGTBG Number-OGTBG Number-Flag
1403-*-Flag
Where,
Flag Meaning
0 Rotation OFF
1 Rotation ON
Step 4
For LCR Type-refer chapter “Configuring LCR”.
Step 5
For CPS refer chapter “Least Cost Routing-Carrier Pre-Selection”.
Step 6
Refer the topic “Station Basic Feature Template” for details on assigning OGTBG to stations.
Step 7
How to assign an access code to OGTBG?
There are maximum 6 trunk access codes. These are Flexible access codes. Trunk access codes are common for
all the users. They cannot be different for different station.
1 0
2 5
3 61
4 62
5 63
6 64
Use following command to program the desirable access code for a trunk access index:
3112-1-OGTBG Index-Access Code-#*/Press <Hold>
3112-2-OGTBG Index-OGTBG Index-Access Code-#*/Press <Hold>
3112-*-Access Code-#*/Press <Hold>
Where,
OGTBG Index is from 1 to 6.
Access Code is maximum 6 digits (Generally access code for trunk is of two digits).
Use following command to clear the access code for a OGTBG index:
3112-1-OGTBG Index-#*
3112-2-OGTBG Index-OGTBG Index-#*
3112-*-#*
Use following command to assign default access code for a OGTBG index:
3162-1-OGTBG Index
3162-2-OGTBG Index-OGTBG Index
3162-*
• OGTBG has “Rotation” flag, and each OG trunk bundle has Rotation type “Cyclic/Descending/Ascending”.
• These two flags don’t have any relation with each other, and so, will work in isolation.
• When there is a second call, it will get routed using the “OG trunk bundle Member 2”, and the trunk port
from the OG trunk bundle member 2 will get selected according to the rotation type programmed in the OG
Trunk Bundle programmed as member2.
• Accordingly, the member of the OGTBG will be accessible to the station user accessing the OGTBG in
sequence.
• Now the trunk/channel to route the call will get selected as per the rotation type programmed for the OG
trunk bundle used as member 2. When the trunk ports of OG trunk bundle programmed in member 1 and
member 2 all are busy, the OG trunk bundle member 3 will be used to route the call.
• Thus the Rotation flag of OGTBG will be used to select the OG trunk bundle member1 to member 8 as per
call basis while the rotation type flag associated with the OG trunk bundle will decide the rotation
mechanism to select the trunk port from the particular OG trunk bundle.
01 None 01 02 03 04 05 00 00 00
02 None 01 00 00 00 00 00 00 00
03 None 01 00 00 00 00 00 00 00
04 None 01 00 00 00 00 00 00 00
05 None 01 00 00 00 00 00 00 00
:: None 01 00 00 00 00 00 00 00
21 None 01 00 00 00 00 00 00 00
22 None 01 00 00 00 00 00 00 00
23 None 01 00 00 00 00 00 00 00
24 None 01 00 00 00 00 00 00 00
25 None 01 00 00 00 00 00 00 00
26 None 01 00 00 00 00 00 00 00
27 None 01 00 00 00 00 00 00 00
28 None 01 00 00 00 00 00 00 00
29 None 01 00 00 00 00 00 00 00
30 None 61 62 63 00 00 00 00 00
31 None 64 00 00 00 00 00 00 00
32 None 61 62 63 64 00 00 00 00
Relevant Topics:
1. “Outgoing Trunk Bundle” 1348
2. “Configuring LCR” 716
3. “Time Tables” 1671
4. “Class of Service (COS)” 1011
5. “Station Basic Feature Template” 436
6. “Trunk Access Group (TAG)” 1692
What's this?
PCAP or packet capture consists of intercepting and logging the traffic passing over a digital network or a part of a
network. PCAP intercepts each packet in the data streams that flow across the network, and can decode and
analyze its contents.
PCAP can be used, among others, to monitor the network, detect and analyze network problems, debug client/
server communications, debug network protocol implementations.
ETERNITY supports PCAP Trace for the Master Ethernet Port, the VoIP Port of ETERNITY, and the Matrix
Extended IP Phones.
Packets traveling over a network are captured and saved in the system. You can save these trace files (packets
captured by the system) on a PC and open these trace files using a graphical packet capture and protocol analysis
tool such as Wireshark or Ethereal.
ETERNITY also supports Filters and 'Promiscuous' mode for capturing packets, which you can use to specify the
types of data packets to be captured.
How to use
• Decide the type of packets to be captured and set the Filter accordingly. The Filter Settings parameter
should be maximum 60 characters in length; all ASCII characters are allowed. By default, this field is
blank. So all packets will be captured.
• To capture packets which are transmitted from the system, from IP address 192.168.1.191:
• Filter Settings = src 192.168.1.191
• To capture only packets which are transmitted from the system and received to the system, IP address
192.168.1.191:
• Filter Settings = src 192.168.1.191 or dst 192.168.1.191
• To capture packets which are transmitted from the system for particular port number only, from IP
address 192.168.1.191 and port number 161 :
• Filter Settings = src 192.168.1.191 and port 161
If you do not enter a valid filter, you will get the message: 'Invalid filter! Please enter valid filter'.
It is not mandatory to set Filters. When the Filter Settings field is left blank, the system will capture all
packets.
• You can set the Enable Promiscuous Mode? to Yes, if you want.
When you enable Promiscuous mode, the ETERNITY will capture all network traffic. However, this will
work only in a non-switched environment.
When Promiscuous Mode flag is disabled, the system will capture only traffic that is directly related to it.
Only traffic to, from or routed through the ETERNITY will be picked up by the PCAP Trace.
'Filter Settings' and 'Promiscuous Mode' (enabled) will not be cleared during power down.
OR
• Wait for the system to stop packet capturing. The system stops packet capturing once the maximum
allotted memory of 1 MB (RAM) is utilized.
• Number of Packets and bytes captured as per the filter setting will be displayed as Packets Captured and
Total Bytes respectively.
Capturing of packets will not stop if you open any other page of Jeeves. So, you may continue using
Jeeves for any other purpose while PCAP Trace is being used.
• When the packet capturing is stopped (by you or the system), click the Save Trace File button to save the
files on the system’s FTP server.
The FTP Login page will open. Enter the SE password. Save your trace file on you local disk.
The current packets captured will not be deleted after you have saved the trace file. The current packets
will be deleted when you start the PCAP capture again.
• Now, you can open the downloaded trace file using Wireshark or Ethereal or any other similar software
which supports opening of trace files.
• Under VoIP Configuration, click the PCAP Trace link to open the page.
• Click the VoIP Port tab for which you want to carry out PCAP Trace. For example, VoIP Port-1.
• Set the Filter for the type of packets to be captured. The Filter Settings parameter should be maximum 60
characters in length; all ASCII characters are allowed. By default, this field is blank. So all packets will be
captured.
You may refer to the following examples for setting the Filters.
src port port number src port 5060 Capture packets if the packet has
a source port value of 5060.
dst port port number dst port 80 Capture packets if the packet has
a destination port value of 80.
port port number port 5060 Capture packets if the packet has either source
or destination port value of 5060.
src host ip address src host 192.168.1.176 Capture packets if the source IP address
is 192.168.1.176
dst host ip address dst host 192.168.1.176 Capture packets if the destination IP address
is 192.168.1.176.
host ip address host 192.168.1.176 Capture packets if either source or destination IP address
is 192.168.1.176
If you enter an invalid filter, you will get an error message: 'Invalid filter! Please enter valid filter' if you do
not enter a valid filter.
It is not mandatory to set Filters. When the Filter Settings field is left blank, the system will capture all
packets.
When you enable Promiscuous mode, the ETERNITY will capture all network traffic. When Promiscuous
Mode flag is disabled, the system will capture only traffic that is directly related to it. Only traffic to, from or
routed through the ETERNITY will be picked up by the PCAP Trace.
'Filter Settings' and 'Promiscuous Mode' (enabled) will not be cleared during power down.
OR
• Wait for the system to stop packet capturing. The system stops packet capturing once the maximum
allotted memory of 2 MB (RAM) is utilized.
Capturing of packets will not be interrupted if you open any other page of Jeeves. So, you may continue
using Jeeves for any other purpose while PCAP Trace is being used.
• When the packet capturing is stopped (by you or the system), you can download the captured file from the
FTP server of the VoIP Card.
The current packets captured will not be deleted after you have saved the trace file. The current packets
will be deleted when you start the PCAP capture again.
• If you want to run PCAP Trace for another port, click the desired VoIP Port Number, and repeat the steps
described above.
• You may open the downloaded trace file using Wireshark or Ethereal or any other similar software which
supports opening of trace files.
You are not required to set any filters; the phone captures all packets that it sends and receives.
To be able to use PCAP Trace for a Matrix Extended IP Phone, there must be a DSS Key for accessing the Local
Menu on the phone. If you have not already configured the DSS Key for Local Menu, you may do so now. For
instructions on configuring DSS keys of the Extended IP Phone, see “Matrix Extended IP Phone Settings” under
Configuring SIP Extensions.
You can access the Local Menu only when your phone is in idle state.
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
CA03 * 0 #
CA02
CA01
LOCAL ME NU
N et wo r k P a r a m e t er s
N et wo r k S t a t u s
N E T W O R K PA R A M E T E R S
M A C :00 :1 b:09 :00 :9a :a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
Sub net Ma sk
G at e w a y A d d r e s s
N E T W O R K PA R A M E T E R S
P P P o E S e rv i c e N a m e
S e r v er A d d re ss
S e r v er P o r t
VLAN S etti ng
PCAP
PCAP T RACE
Sta rt PCAP
PCAP TRACE
Sta rted !
To stop PCAP,
• Enter the Local Menu of the phone again by pressing the DSS key.
PCAP T RACE
Stop PCAP
PCAP T RACE
Sto ppe d!
• Go idle.
You can download the Trace file from the embedded FTP server of the Extended IP Phone. To access the FTP
server using Windows FTP, do the following:
• Go to My Computer.
• Type the current IP Address of the Extended IP Phone in the Address bar. For example: ftp://
192.168.201.134
• In Password, enter the current User Password set for the phone.
• On successful login, the FTP window will open. You will see the different Configuration folders in this
window.
• In the ramdisk folder, right click the file trace.pcap and copy it on to your local disk.
• Open the trace.pcap file using Wireshark or Ethereal or any other similar software which supports
opening of trace files.
You may also use FireFTP, if you are using Mozilla Fire Fox. Make sure your browser has the FireFTP
Add-on installed.
What's this?
Paging allows you to make announcements to groups of extension users and to make public announcements over
a public address system. You can deliver a message to a mass of people at once by just lifting the handset of your
phone and dialing a code.
This feature is useful when you want to call several people at once; for example, to inform them about a meeting
you have scheduled. If the persons you want to call have Digital Key Phones (DKP) as their extensions, you can
use paging instead of calling them up one by one.
This feature is of great use in factories, offices, where it is not feasible to provide individual extensions in every
place, or when announcements are to be made in a hall, a lobby or a shop floor. In such situations, if you need to
call or inform someone, you can simply make an announcement over a Public Address System (PAS).
• External Paging: Announcements are made on a Public Address System (PAS) connected to the Analog
Output Port of the ETERNITY.
For both types of Paging, the DKP extensions which are to be paged and the Analog Output Port to which the PAS
is connected must be included in 'Page Zones'.
• You can page from both SLT and DKP extensions, but the extensions you page to must be only DKP
extensions.
• The DKP extension being paged can only hear the person who is paging as the DKP speakerphone
mic is not activated.
How it works
The Pre-requisites
• Page Zones must be created. Each Page Zone accommodates up to 16 DKP extensions and the Analog
Output Port. You can create 12 different Page Zones of 16 DKP extensions each.
• Paging must be enabled in the Class of Service allowed to the DKP/SLT extension from which this feature
is to be used.
• For External Paging, a Public Address System must be connected to the Analog Output Port of the
ETERNITY. For instructions, refer the topic 'Connecting a Public Address System' under “Installing
ETERNITY ME”, “Installing ETERNITY GE” and “Installing ETERNITY PE”186.
• The system activates the speakerphones of the DKPs programmed in the Page Zone number. The system
activates the speakerphones only those DKP ports in the Page Zone that are free.
If the Analog Output Port is included in the Page Zone the PAS connected to this port will be activated.
• All DKPs in the Page Zone can hear the announcement. But they cannot speak to the calling DKP/SLT
extension user (as the mic of their speakerphones is not activated).
If the Analog Output Port is included in this Page Zone, the announcement will be heard by the public.
• The system deactivates the speakerphones of the DKPs and the PAS connected to the Analog Output
Port.
How to configure
For this feature to work, you must create Page Zones and enable this feature in the Class of Service of the
extensions which are to be allowed this feature.
If you want to allow Paging to all stations, retain CoS group 01 in Template 01.
b. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
For each Page Zone number, decide and assign the DKP extensions and the Analog Output Port.
On a sheet of paper create a two-column table for each page zone, as shown below. Enter the Software Port
number of the DKP extensions you want to include in the page zone. Also specify if you want to use the Analog
Output Port in a Page Zone.
Page Zone 1
1 002
2 003
3 008
4 009
16
Page Zone 2
1 007
2 010
3 012
4 013
16
If you want to include only the Analog Output Port in a Page Zone, you do not need to assign any DKP extensions
to that Page Zone.
Make a note of the Page Zone for which you want to use only the Analog Output Port.
Now, program Page Zones using Jeeves or by dialing SE commands from a Telephone.
• Refer to the Page Zones you created on paper and program the following parameters in each Page Zone:
• Page on AOP: Select this check box if you want to include AOP in the Page Zone.
If you want to include only the AOP in a Page Zone, do not program any of the DKPs in the Page Zone.
• DKP Port: Enter the Software Port number of the DKP you want to include in the Page Zone.
• Include in Page Zone: Select the check box to include the DKP Port number you entered as member
of the Page Zone.
• Click 'Submit' at the bottom of the page to save your Page Zone settings.
• To go to other Page Zones, you may click the hyperlinked page zone numbers on the top of the Jeeves
screen.
• If you have completed programming Page Zones, you may log out of Jeeves.
• Exit SE mode.
How to use
It is possible to page from a DKP as well as SLT.
What’s this?
• Making a call on the VoIP Ethernet Port without going through any proxy is called Peer-to-Peer Calling.
• For this the system supports, the table to be programmed which consists of Number, Destination Address
and Name field for each entry.
• The user has to know only the IP address of the called party.
For programming ‘.’ use ‘#9’ when SLT or DKP is used. But for dialing a number with ‘.’ use ‘*’.
For example:
There are 3 branch offices of the same company located at different places and they are connected to each other
by private IP Network as shown in the figure below:
Baroda Mumbai
Frankfurt Chicago
PBX-B PBX-D
[email protected] [email protected]
24251 24531 651011 890011
192.168.1.3
Delhi
PSTN
11
31
PBX-C
9898001122 6545351
2001 2002
Following types of calls can be made using peer to peer calling feature.
• 2001 will dial ‘3001’ to call Mumbai office.
• 2001 can dial 31-2001 to call Delhi office, i.e with PBX exchange ID.
• 2001 can use trunk Access Code ‘0’ to call Mumbai office. He can dial ‘0-3001’.
• 2001 can call 2001 in Delhi by dialing Trunk Access Code i.e. by dialing 0-312001.
• Baroda local user 2630555 can call the Baroda user through Baroda-PBX i.e 2630555 user will dial any
PSTN number of PBX for which DISA is enabled and dials the Chicago number using TAC i.e. 0011-312-
651011.
• 2001 can transfer the call to 3001 in Mumbai (whether Internal or External Call).
• 2001 can call [email protected] directly through SLT or DKP using TAC.
• 2001 can call IP address 192.167.100.1 directly through SLT or DKP using TAC.
• [email protected] and IP address 192.167.100.1 can be stored in the Global Memory/Personal Memory.
Toll control will be checked for IP Address.
Above mentioned different applications of the non-proxy calling are possible with suitable programming of the
ETERNITY with ETERNITY ME Card VoIP. Programming for different applications of calling is explained in detailed
in Important points.
How it works
Peer to Peer Table is used, for deciding the destination IP Address for call routing on the non-proxy SIP trunks
(Registrar server address shall be programmed as blank for these trunks). This table is common for all the SIP
trunks.
• Program three parameters viz. Number String Prefix, Destination Address and Name for each entry.
• Number String Prefix can be maximum 8 digits long. (0-9, #, * allowed). Default: Blank.
• Program the first entry for destination Address and Name which will be used to route the call in case the
dialed number is not found in the Peer-to-Peer call table. Thus entries at Index-001 are reserved for the
case of dialed number string not programmed in the Peer-to-Peer Table.
• After this table, the system will check Automatic Number Translation (ANT) logic for dialing the number.
• Programming of the Peer-to-Peer Table can be done through Web Jeeves and SE mode using DKP or
SLT.
How to configure
The commands explained below should be referred as:
To program a single port: XXXX-1
To program a range of ports: XXXX-2
To program all the ports: XXXX-*
Number String
Destination Address
Name
Display Name field is not sent in the SIP message. It is just a tag for the entry.
• Refer the chapter “Configuring SIP Trunks”. The system will consider the SIP trunk to be party, if the
registrar server is programmed for it. If the registrar server address is not programmed, it will be
considered as non-proxy trunk.
• Using existing CUG table for making outgoing call using SIP Trunk:
• We can use the existing CUG Table of the system i.e. program the extension number or PBX
Exchange ID in the CUG table and Assign the OGTBG (consisting of SIP Non-proxy trunk).
• Thus when the user dials the extension number directly or with PBX exchange ID, CUG table will be
checked first and if the number is found in the CUG table, call will be routed using the OGTBG
programmed for that number.
How to configure the PBX and how the call will be routed when different
type of calls is made is explained below:
Case 1:
• Program 3001(or 3 or 30 or 300) with OGTBG containing SIP Trunk (Non-Proxy Trunk) in the CUG
table of PBX-A. Program Maximum dialed digits as 4 for this index. All other parameters in the CUG
table should be blank.
• Program 3001(or 3 or 30 or 300) and Destination Address 192.168.1.1 in the Peer-to-Peer table.
• When 2001 dials 3001, it will be compared in the CUG table. As number is matched in the CUG table,
OGTBG is decided.
• From OGTBG it is decided that it is a SIP trunk, thus Peer-to-Peer table is checked and Destination
address is found from the table.
• Use the ‘SIP’ as station for incoming call, by assigning ‘*’ as ‘SIP ID’ for the SIP trunk.
• PBX-E checks the CUG table first and as 3001 is not programmed in the CUG table, it checks the
flexible number of the stations.
• The call will be routed even if 3001 is programmed in the CUG table. For this, enable self route flag.
Case 2:
2001 of PBX-A dials 31-2001 (with PBX exchange ID) after picking up the handset. The call should land on
station 2001 of PBX-C:
• Program 31(or 312 or 3120 or 312001) with OGTBG containing SIP Trunk (Non-Proxy Trunk) in the CUG
table of PBX-A. Program Maximum dialed digits as 6 for this index. All other parameters in the CUG table
should be blank.
• Program 31(or 312 or 3120 or 312001) and Destination Address 192.168.1.3 in the Peer-to-Peer table.
• Program 31 (or 312 or 3120 or 312001) with Self Route Flag Enabled and strip digit count2 in the CUG
table of PBX-C.
• Use the ‘SIP’ as station for incoming call, by assigning ‘*’ as ‘SIP ID’ for the SIP trunk.
• When 2001 dials 312001, it will be compared in the CUG table. As number is matched in the CUG table,
OGTBG is decided.
• From OGTBG it is decided that it is a SIP trunk, thus Peer-to-Peer table is checked and Destination
address is found from the table.
• PBX-C checks the CUG table first and 312001 is found in the CUG table with self route flag enable and
strip digit count 2.
• Thus the call is routed to the station 2001 of PBX-C after striping of 31.
Case 3:
2001 of PBX-A dials 3001 using Trunk Access Code (TAC) after picking up the handset i.e. by dialing 0-3001:
• Program LCR-Number Based (assign OGTBG containing SIP trunk (Non-proxy) to number 3) or assign
SIP Trunk (Non-Proxy trunk) for the TAC 0.
• Program 3001(or 3 or 30 or 300) and Destination Address 192.168.1.1 in the Peer-to-Peer table.
• Use the ‘SIP’ as station for incoming call, by assigning ‘*’ as ‘SIP ID’ for the SIP trunk.
• From OGTBG it is decided that it is a SIP trunk, thus Peer-to-Peer table is checked and Destination
address is found from the table.
• PBX-E checks the CUG table first and as 3001 is not programmed in the CUG table, it checks the flexible
number of the stations.
Case 4:
2001 of PBX-A dials 2001using Trunk Access Code i.e. by dialing 0-312001. Call should land on station 2001
of PBX-C:
• Program LCR-Number Based (assign OGTBG containing SIP trunk (Non-proxy) to number 31) or assign
SIP Trunk (Non-Proxy trunk) for the TAC 0.
• Program 31(or 312 or 3120 or 312001) and Destination Address 192.168.1.3 in the Peer-to-Peer table.
• Use the ‘SIP’ as station for incoming call, by assigning ‘*’ as ‘SIP ID’ for the SIP trunk.
• Program 31 (or 312 or 3120 or 312001) with Self Route Flag Enabled and strip digit count2 in the CUG
table of PBX-C.
• When 2001 dials 0-312001, OGTBG is decided as per the programming i.e. LCR-Number based or
OGTBG assigned to TAC.
• From OGTBG it is decided that it is a SIP trunk, thus Peer-to-Peer table is checked and Destination
address is found from the table.
• PBX-C checks the CUG table first and 312001 is found in the CUG table with self route flag enable and
strip digit count 2.
• Thus the call is routed to the station 2001 of PBX-C after striping of 31.
Case 5:
2001 should be able to call the local PSTN number 28111263 of Mumbai by dialing Trunk Access Code i.e. by
dialing 0-02228111263. The call should be routed to the PSTN number 28122263 of PBX-E:
• Program LCR-Number Based (assign OGTBG containing SIP trunk (Non-proxy) to number 022) or assign
SIP Trunk (Non-Proxy trunk) for the TAC 0.
• Use the ‘SIP’ as station for incoming call, by assigning ‘*’ as ‘SIP ID’ for the SIP trunk.
• Program 022 with OGTBG containing any analog or digital trunk in the CUG table of PBX-E. Program Strip
digit count = 3. All other parameters in the CUG table should be blank.
• When 2001 dials 0-02228111263, OGTBG is decided as per the programming i.e. LCR-Number based or
OGTBG assigned to TAC.
• From OGTBG it is decided that it is a SIP trunk, thus Peer-to-Peer table is checked and Destination
address is found from the table.
• PBX-E checks the CUG table first and as 022 is programmed in the CUG table, it decides the OGTBG.
• Using the trunk programmed in the OGTBG, it dials out the number 28111263 after stripping off the digit
022.
• Thus the call is routed to the PSTN number 28111263 through PBX-E.
Relevant Topics:
1. “Closed User Group (CUG)”
2. “Configuring SIP Trunks”
What’s this?
• ETERNITY-PLCC EPAX is a digital PBX. It uses a digital switch and hence is a 100% non-blocking
system. In PLCC network, number of PLCC EPAX needs to be connected.
• Refer to diagram showing cluster of exchanges in a PLCC network. Each exchange in PLCC network is
assigned with Exchange Identity (SID) in order to get identified by other exchanges in the network. Thus
each exchange is identified by Exchange Identity (SID). As shown, an exchange is connected to the other
exchange through an E&M tie line. Also, it is possible that an exchange may not be directly connected to
all other exchanges in a PLCC network through E&M tie lines.
• For example, SID-52 exchange is connected to SID-51, SID-34, SID-71 and SID-60 exchanges through
direct E&M tie lines. However, SID-52 exchange is not directly connected to SID-61, SID-62 and SID-72
exchanges through E&M tie lines.
• Consider, an example where subscriber 20 of SID-52 needs to call subscriber 30 of SID-51. After
dialing trunk access code, dial 51 (SID number of the exchange) and then dial 30 (subscriber number
of SID-51). This is how a call is completed, as both exchanges; SID-51 and 52 are connected through a
direct E&M tie line.
• However, consider another example, where subscriber 20 of SID-52 needs to call subscriber 10 of SID-
61. Even though, here both exchanges are not connected through direct E&M tie line, it is possible by
dialing trunk access code with SID number of the exchange (here 61) followed by subscriber number
(here 10). However, the call will proceed through SID-60 and then reach SID-61. This is called Transit
Call. Here, the subscriber will not be able to know that call has proceeded to the required exchange
through transit facility.
PLCC Network
Sub. 10
SID-61 Sub. 11
Sub. 12
SID-34 SID-60
Sub. 20
SID-52 Sub. 21
Sub. 22
SID-51 SID-71
SID-72
Sub. 30 Sub. 31
PLCC-EPAX (SID-61)
E&M1 E&M2 E&M3 E&Mn
PLCC-EPAX
(SID-70)
E&M1 E&Mn
PLCC-EPAX E&M1
(SID-34)
E&M1 E&M2
E1 PLCC-EPAX E1
E&M2 (SID-52)
E2 E2
E&M3 E&M3
En En
E&Mn E&Mn
PLCC-EPAX (SID-51)
E1 E2 En
PLCC-Priority
• All callers do not have same hierarchical position in an organization. It is not advisable to keep a call
waiting ofsome important person just because there is already one unimportant call pending at the
destination. The important caller should be allowed to jump the queue and be attended first ahead of other
earlier pending calls.
• The ETERNITY supports flexible priority assignment for different users. Each port can be assigned a
priority level between “0” to “9”. Higher the priority level, more important the caller is. Accordingly 0 has the
least priority and 9 has the highest priority.
• This feature enables a station user to free the system resources (station or a trunk) for him.
The PLCC functional module requires a license. You must purchase a license to activate CCS Signaling
when End Point and Transit, Express Signaling, and Seizure Pulse and Release Pulse Signaling. Refer the
topic “License Management”.
The called station/trunk gets disconnected. You get dial tone after the
3 Dial #*.
confirmation tone.
How it works
• Suppose, Station A and Station B are talking to each other. Station C calls Station B and finds it to be busy.
If he uses Priority, he gets connected to Station B. Station A gets disconnected and gets error tone.
However, for this to happen the priority of Station C should be higher than that of Station B and Station A
else he won’t be able to use this feature.
• Likewise suppose Station A is talking to an external party through trunk 1. Station B tries to grab Trunk 1.
On finding it busy, he uses Priority. Doing so, he gets connected to trunk 1 and gets P&T dial tone whereas
Station A gets disconnected and gets error tone. However, for this to happen priority of Station B should
be higher than Station A. In this case, priority of the trunk is not considered.
• Suppose station A is talking to an external party through trunk1. Station B calls station A and finds it busy.
He uses Priority. Doing so, station B gets dial tone whereas, the trunk 1 gets disconnected.
How to use
Please refer above diagram. If any two subscribers, say 20 and 21 of an exchange with SID-52 (without priority
access) are in conversation and a third subscriber, say 22 (of same exchange) with priority access and wants to
talk with subscriber 21. But, when subscriber 22 dials for subscriber 21, it gets busy tone. Subscriber 22 can use its
priority by dialing code ‘#*’ in such case and can come in conference with subscribers 20 and 21.
How to use
Please refer above diagram. If any two subscribers, say 20 and 21 of an exchange with SID-52 are in conversation
and a subscriber, say 10 of an exchange with SID-61 (from the same network with priority access) want to talk with
subscriber 20 of SID-52. After dialing the required subscriber 20 of SID-52, he gets busy tone. Subscriber 10 of
SID-61 can now press ‘#*’ on his telephone and can get into conference with both subscriber 20 and 21 of SID-52.
In conversation
How to use
Please refer diagram, if two subscribers, say 20 of SID-52 (without priority access) and 11 of SID-61 (without
priority access) are in conversation on an E&M tie line. Both this exchanges SID-52 and SID-61 are connected with
each other through a single E&M tie line. Now, if subscriber, say 10 (with priority access) of SID-61 wants to talk
with subscriber 21 of SID-52, then he will get busy tone. Subscriber 10 of SID-61 can now use its priority by
PLCC-Routing Table
• In PLCC network, number of PLCC-EPAX needs to be connected. The entire network should behave as a
single unit or one group. It is not feasible to have unique station numbers throughout the network. In such
cases, an Exchange ID is assigned to the PBX and a routing table is programmed in the exchange. In case
of newly added exchange in the network, the routing tables of other exchanges required to be modified.
In the above figure, 3 PBX systems are connected through E&M connectivity.
How it works
In this application, it is possible to have same station number in two or more PBXs of the network. Few new words
have been used to explain PLCC routing table, each of these words have been explained below:
Routing Table: This table has five parameters viz. Route Index, Route Code, OG Trunk Bundle Group, Strip Digit
Count and Self Route flag. The PLCC routing table programming works according to this table.
001
250
Route Code: Route code could be of maximum sixteen digits. Digits 0 to 9, # and * are allowed. Generally, route
code should be a unique number. For example in the figure given above, route code for PBX-A can be defined as
‘21’, route code for PBX-B can be defined as ‘22’ and that for PBX-C can be defined as ‘23’.
OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code. Whenever a call
is to be made on that route, a free trunk from the OGTBG is selected and the station number is dialed on it. The
same logic of rotation On/Off for trunk selection from the OGTBG is used. If rotation is OFF then always the first
trunk in the OGTBG is selected. If it is busy then the next trunk in the group is selected. This helps to select an
alternate route. Whereas if rotation is ON then the trunks in the OGTBG are selected in round robin fashion.
Strip Digit Count: This count signifies the number of digits to be striped of while dialing/decoding a number. To
elaborate: Consider figure 1. The requirement is that if station 2001 of PBX-B dials 212002 and if E&M 1 is busy
then the call should reach station 2002 of PBX-A through alternate route. In this case the strip digit count of PBX-A
should be programmed as 2 and that of PBX-B and PBX-C should be programmed as 0. Doing so, when station
2001 of PBX-B dials 212002 and if E&M1 is busy then the call is routed through PBX-C. In this case, PBX-B dials
212002 on E&M3, PBX-C receive this code and dials out the same code i.e. 212002 on E&M2 without striping of
any digit. On receiving 212002, PBX-A strips of two digits as per the programming and routes the call to station
2002.
Self-Route Flag: This flag signifies that the digits being dialed are for the same PBX and are not to be dialed on the
E&M trunk.
Maximum dialed digits: When digits are dialed on the trunk, the system waits for inter digit timer after the last digit
is dialed. In order to avoid this timer and number of digits dialed to be routed without further delay, count for the
number of digits to be programmed in this field. If the number of digits received are equal to the parameters
programmed then the number is dialed out immediately without waiting for the inter digit timer. If the number of
digits dialed by the user are not equal to the digits programmed, the number is dialed after inter digit timer.
Consider a case in which PLCC network has exchanges ‘A’, ‘B’, ‘C’ with station numbers 2001 to 2100 for PBX ‘A’,
3001 to 3100 for PBX ‘B’ and 4001 to 4100 for PBX ‘C’. Route code can be defined as ‘2’, ‘3’ and ‘4’ for PBX ‘A’,
PBX ‘B’ and PBX ‘C’ respectively. When station number 2001 wants to talk to 2050, he dials the required number.
PBX ‘A’ checks subsequent dialed digit and transfer the call to station number 2050. If first digit happens to be 3 or
4 the call is transferred to PBX ‘B’ or PBX ‘C’ respectively. PBX ‘B’ and PBX ‘C’ checks for second and subsequent
digits. If the dialed number is available in the exchange then the call is transferred to destination. If the dialed
number is not available then the error message will be given by that particular exchange.
The PLCC Routing table should be programmed carefully. Specified Route code in association with Self Route Flag
disable will make the call to be routed on trunk specified in OG Trunk Bundle Group.
The ETERNITY has only one routing table. The same table is used for Closed User Group and Closed
User Group-With Exchange ID. Hence the table has to be programmed keeping the application in mind.
Refer to diagram in ‘Introduction to PLCC Network’ topic showing cluster of substations in a PLCC network. A
substation in a PLCC network is connected to any other substation through a dedicated PLCC express line (i.e. 4/6
wire E&M tie line) as shown in the diagram.
How to configure
PLCC Express Line Communication System is one of the applications of the ETERNITY. We recommend the user
to read relevant topics before programming it for PLCC Express Line application.
Step 1
Refer chapter “E&M Feature Template” to program the feature in an E&M Feature Template.
Step 2
Refer chapter “E&M Feature Template” to assign default values to an E&M Feature Template.
Step 3
Refer chapter “E&M Feature Template” to assign an E&M Feature Template to an E&M.
Step 4
Refer chapter “Configuring DKP Extensions” to program a name for E&M trunk.
Step 5
Hardware ID is an attribute of a software port. Hardware ID of a software port decides where the port is physically
located. To derive hardware ID of a software port, we need slot number and port number of the card. Hence, all the
programming is done for the software port and not for the hardware ID. Accordingly, the software port number is
used for all the programming. Please refer “Software Port and Hardware ID” for more details, to assign hardware ID
to an E&M software port:
Step 6
Refer chapter “Station Advanced Feature Template” to assign a Station Advanced Feature Template to an E&M.
Step 7
Refer chapter “Station Advanced Feature Template” to assign a Station Basic Feature Template to an E&M.
Step 8
Please refer “Time Tables” chapter to program the time zone of a trunk.
Step 9
To program a feature in a Trunk Feature Template, please refer “Trunk Feature Template” for more details.
Step 10
To assign a Trunk Feature Template to an E&M, please refer “Trunk Feature Template” topic.
Function:
A function should be assigned to these Keys, which they should perform. There are 2 different types of functions,
which can be assigned to a key for PLCC express line application. Each function that can be assigned to a key is
given unique number. Following table list the function types available:
00 Null --
Step 12
Refer chapter “DSS Keys Programming” to assign a function to a DKP key. In this command refer above
explanation to program ‘Function Type’.
Step 13
Define dialing property of E&M port. The dialing properties are based on the applications. If the dialing type for a
E&M is programmed as pulse type then Pulse-Dialing ratio should be defined. The E&M support six different Pulse
Dialing Ratio’s.
Value Meaning/Ratio
1 10PPS, 1:2
2 10PPS, 2:3
3 10PPS, 1:1
4 20PPS, 1:2
5 20PPS, 2:3
6 20PPS, 1:1
For assigning Dial Type and Pulse Dial Ratio, please refer “E&M Feature Template” topic.
Relevant Topics:
1. “Trunk Feature Template” 546
2. “Time Tables” 1671
3. “E&M Feature Template” 565
4. “Closed User Group (CUG)” 1028
5. “Closed User Group-With Exchange ID” 1033
What's this?
Extension users may want to indicate their availability to callers from other extensions.
For example, an extension user may want to leave his desk for an indefinite period, but does not want to use Call
Forward or set Do Not Disturb. He wants to indicate to callers about his absence. Similarly, extension users who
are present at their desk may want to hide their presence from other users; or they may want to show their current
activity to the other extension users like they are Busy, or are away from their desks, or on the phone with someone
on another call, etc.
With the Presence feature of ETERNITY, extension users, including the Operator, can 'publish' their presence to
callers from other extensions. By doing so, they can indicate to the other extensions about their availability.
In the same way, the Presence feature allows extension users to view the 'Presence' status (availability) of the
extensions that they want to call, before making the call or when their call is not answered.
How it works
Publishing Presence
Any SLT, DKP, ISDN Terminal, and SIP Extension User can 'publish' their presence by setting any of the messages
listed in the following on their phone, by dialing the access code for this feature.
SIP Extension users who want to publish their presence have two options:
• Using the PUBLISH feature supported by the SIP Client.
• Using the feature access code for Publish Presence supported by ETERNITY.
The first option requires the parameter 'PUBLISH' to be enabled in the SIP Extension Settings. Refer
“Configuring SIP Extensions”. By default, this parameter is disabled.
When any other DKP extension user calls this extension, the text message 'User Absent' will appear on
the caller's phone display.
If the extension phone that has set 'Absent' is a DKP, the letter 'A' appears on the phone's display to
indicate absence.
The letter 'A' disappears when the extension user sets a presence message other than 'Absent'.
As 'Absent blocks incoming external as well as internal calls, it can be used as an extension of the “Do Not
Disturb (DND)” feature, which blocks only internal incoming calls.
• External callers who call the extension, on which 'Absent' is set, will get an error tone only.
• Outgoing calls can be made from the extension which has set 'Absent'. Only incoming calls are
restricted.
• If more than one extension is configured as "Operator" (routing group), incoming calls will be blocked
only on the Operator extension which has set User Absent.
1. Present:
When an extension user sets 'Present', all incoming calls will be received as normal on this extension.
If previously set as 'Absent', when a DKP extension user sets 'Present' the letter 'A' will disappear from the
phone's display.
When any other DKP extension user calls this extension, the name of the extension user will be displayed
on the caller's phone display, when the called extension is ringing.
2. Auto Detect: When an extension user sets 'Auto Detect', the system will detect the state of the phone;
depending on the call state, it will publish the presence message to the other extensions. Three types
Publish Presence messages are possible, with Auto Detect:
a. Idle: When the system detects the extension phone to be ON-Hook, it indicates the status of the phone
to other extensions 'idle'.
b. On the Phone: When the system detects the phone to be OFF-Hook, or in speech with another party
or if it detects an incoming call placed on the phone, it will indicate to the other extensions that this
extension user is 'On the Phone' with another party.
c. DND Text message: When the system detects that the extension phone has Do Not Disturb (DND) set
on it with a DND Text message, it will display to the calling extension, the DND message set by the
called party (this may be the default DND message or the DND Text message set by the called
extension).
3. Away: When an extension user sets 'Away', the system will display this message to the other extensions.
4. On the Phone: When an extension user sets 'On the Phone', the system will display this message to the
other extensions.
5. Do Not Disturb: The extension user can set this message to be published to other extensions, if s/he
wants to work uninterrupted.
Unlike the DND Feature, the extension user who has set this message will continue to receive calls both
internal as well as external calls, as the system considers this extension as 'present'.
6. I am Mobile: The extension user can set this message to be displayed to other extensions, when s/he is
not at the desk.
8. Out for Meal: The extension user can set this message to be displayed to other extensions when going on
a lunch break.
9. Out of Office: The extension user can set this message to be displayed to the callers when s/he leaves
the office temporarily.
When an extension user sets any Publish Presence message other than 'Absent', the system will consider
the user as 'Present'. All incoming and outgoing calls will be allowed on this extension.
It is possible to program another message in place of Publish Presence messages listed from 6 to 9: I am
Mobile, In a Meeting, Out for Meal, Out of Office.
Publish Presence messages can be set or changed for any extension from the System Administrator (SA)
mode.
Viewing Presence
• Extension users can know the status of another extension user before calling or when the extension user
does not answer the call.
• Generally, when DKP extension users call another extension, the name of the called extension is
displayed on the calling DKP extension. Now, if the flag 'Display User Status during Call' is enabled in the
System Parameters, when DKP extension users call another extension, the calling DKP extensions will be
displayed the 'presence' status message published by the called extension187.
• SLT extension users, whose phone is equipped with a CLI display, can see the status of another extension
by dialing a feature access code, then going ON-Hook. The system will ring back the SLT and send the
Presence status of the desired extension as CLI.
• SIP extension users can use the Presence feature of ETERNITY to view the presence status of other
extensions. For this, they must dial the feature access code and the number of the desired extension.
• SIP extension users who want to view the status of other extensions using the feature supported by their
SIP Client, must have 'Presence Subscription' enabled in their SIP Extension Settings. Refer “Configuring
SIP Extensions”.
How to configure
This feature involves the programming of the following parameters:
• 'Display User Status during Call' flag: DKP extension users will be able to view the presence status for
the called extension only if this flag is enabled in the System Parameters.
• PUBLISH: SIP extension users who want to publish their presence using the feature supported by their
SIP client will be able to publish their presence status only if this feature is enabled in their SIP Extension
Settings. This parameter is not necessary, if they want to publish presence using the feature of ETERNITY.
187. DKP users can also dial a feature access code and the number of the extension to see the status of that extension on their DKP.
But this would not be required, if the 'Display User Status during Call' flag is enabled in the System Parameters.
• Publish Messages: It is possible to customize the Publish Messages listed above from 6 to 9 viz.: 'I am
Mobile', 'In Meeting', 'Out for Meal', 'Out of Office'.
The above parameters, with the exception of 'Publish Messages', can be programmed using both Jeeves and
a Telephone. You can program 'Publish Messages' using Jeeves only.
• Go to 'Display Presence Status during Call on DKP'. Click to enable the flag.
• You can change message number 6 to 9 as desired. The string may consist of a maximum of 16
characters.
• Click 'SIP Extension Settings' under VoIP Configuration to open the page.
• Now, go to the desired SIP Extension number for which you want to enable the features PUBLISH and
Presence Subscription. By default both features are disabled. Click the respective check boxes to enable
the features.
• Exit SE mode.
How to use
This feature requires you to dial your User Password. The default User Password 1111 is not accepted. Please
change the User Password first.
Publish Presence can be set for an extension also from the System Administrator mode.
OR
• Dial 104
• Enter User Password on the prompt.
• Scroll to the desired Publish message from the menu:
• Absent
• Present
• Auto Detect
• Away
• On the Phone
• Do Not Disturb
• I am Mobile
• In Meeting
• Out for Meal
• Out of Office
• Press Enter key to select message.
• You get the confirmatory tone.
OR
• Dial 1072-014
• Enter Destination Number, i.e. the number of the extension Publish Presence is to be set.
• Scroll to the desired Publish message from the menu:
• Absent
• Present
• Auto Detect
• Away
• On the Phone
• Do Not Disturb
• I am Mobile
• In Meeting
OR
• Dial 1097.
• Enter Extension number
• The status of the extension number you dialed will be displayed on your phone's LCD.
• Go ON-Hook.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
9 Out of Office
• Replace handset.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
9 Out of Office
• Replace handset.
• Lift handset.
• Dial 1097-Extension Number.
• You get confirmation tone.
• Go ON-Hook during confirmation tone.
• Your phone will ring and the status of the extension number you dialed will be displayed on your phone as
CLI.
What's this?
In a network of PBXs connected over E&M trunks, callers may be unable to reach the desired station, as the called
station may be busy with another station.
The problem becomes more acute when important decision makers and persons involved in critical tasks in an
organization are unable to get through to the desired extension of the networked PBX, because it is busy.
ETERNITY supports Priority Call in E&M MFCR2 Signaling to overcome this. This feature makes it possible to land
a call on a busy extension and speak to the extension user. If you required, you can disconnect the remote party in
speech with the called extension.
E&M MFCR2 Signaling is currently supported on ETERNITY GE E&M Card only. This feature is currently
available on ETERNITY GE only.
How it works
• The extensions which are to be allowed Priority Call feature must be defined as 'Priority Subscriber' in their
Station Advanced Feature Templates.
• The extensions on which Priority Calls should be allowed to land must be 'Non-Priority Subscribers' (i.e.
defined as 'Ordinary Subscriber' in the Station Advanced Feature Template).
• In the following illustration, ETERNITY is networked with PBX-A over E&M Lines with MFCR2 Signaling.
2001 3001
M M
E E
Rx Rx
ETERNITY PBX-A
Tx Tx
SA SA
2002 3002
SB SB
• Extension 2001 of ETERNITY is a 'Priority Subscriber'. Extension 2002 is an 'Ordinary Subscriber' (Non-
Priority).
• Extension 3001 of PBX-A is a 'Non-Priority Subscriber'. Extension 3002 of PBX-A is also a 'Non-Priority
Subscriber'.
• If Extension 3001 is busy with another extension, 3002, and since both are non-priority subscribers, PBX-
A will automatically intrude the conversation between 3001 and 3002 and establish 3-way speech between
Extensions 2001, 3001 and 3002.
• Now, if Extension 2001 (the Priority caller) does not want Extension 3002 to be part of the conversation, s/
he can ask 3002 to go idle OR s/he can use the feature Forced Release Order to disconnect 3002 from the
conversation. 2001 can use Forced Release Order only if it is allowed to it in its “Class of Service (COS)”.
• When Extension 2001 dials the Feature Access Code for Forced Release Order, Extension 3002 will be
disconnected and two-way speech will be established between 2001 and 3001.
3001 2001
M M
E E
Rx Rx
PBX-A ETERNITY
Tx Tx
SA SA
3002 2002
SB SB
• Extension 3002 of PBX-A is a Priority Subscriber. Extension 3001 of PBX-A is Non-Priority Subscriber.
• Extension 2002 is busy with Extension 2001. ETERNITY checks the priority of both extensions. Since both
are non-priority subscribers, ETERNITY gives priority to Extension 3002 and plays beeps to 2002 and
2001 before establishing 3-way speech.
• If the busy call is itself a priority call, i.e. at least one of the two parties in the busy call is a priority
subscriber.
• If any party involved in a matured call is Priority Extension or Trunk, the call will be considered as
priority call and hence no incoming priority call will be allowed to intrude this priority call.
• ETERNITY allows non-priority subscribers to intrude on a busy call using the feature Manual Priority
Intrusion. For this, the intruding extension must have the feature 'Manual Priority Intrusion' in its Class of
Service.
• If intrusion is possible ETERNITY creates a conference call, after playing beeps to notify both parties of the
3-way conference call. This beep will be played only if Conference beeps are enabled.
How to configure
For this feature to work, the extensions of the networked PBXs which are to be allowed Priority Call feature must be
defined as 'Priority Subscribers'.
Extensions on which Priority Calls must be allowed to land may be defined as 'Non-Priority Subscribers'.
If non-priority subscribers are to be allowed to Priority Calls when necessary, Manual Priority Intrusion feature must
be enabled in their Class of Service.
If the Priority Subscriber extension users are to be allowed to disconnect the second party (other than the desired
party) from the conversation during a Priority Call, the feature Forced Release Order must be enabled in their Class
of Service.
For Stations on which data terminals are connected (like Fax machine), it is recommended that the 'Station
Category' of these stations be programmed as 'Data Transmission' (in their Station Advanced Feature
Template).
In the default Station Advanced Feature Template 01 assigned to all extensions of the ETERNITY, the Station
Category is set to 'Ordinary Subscriber', which means all extensions are by default, 'Non-Priority Subscribers'.
Since all extensions cannot be defined as 'Priority Subscribers', prepare a new Station Advanced Feature
Template, select the option 'Priority Subscriber' as Station Category in this template and apply it on the selected
extension that are to be defined as Priority Subscribers.
Refer the topic “Station Advanced Feature Template” for detailed instructions.
Programming Manual Priority Intrusion and Forced Release Order in Class of Service
In the default the default CoS group 01 in Station Basic Feature Template Number 01 assigned by default to all
extensions of ETERNITY, 'Manual Priority Intrusion' and 'Forced Release Order' are disabled.
If you want to allow all extensions these features, simply enable them in the default CoS group 01.
However, if either or both these features are to be allowed only to select extensions, follow these steps:
1. Define a CoS group with Manual Priority Intrusion/Forced Release Order enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions.
How to use
What’s this?
‘Priority’ is the precedence given to certain trunks and extensions over others in being answered by the destination
extension.
When ‘Priority’ is assigned to trunks, whenever there are incoming calls on multiple trunks at the same time, the call
on the trunk with higher priority will be answered by the landing destination extension/Operator first.
When Priority is assigned to Extensions, calls from extensions with higher priority will have precedence in landing
on the destination extension.
Higher Priority can be assigned to Extensions of important or higher ranking persons in an organization; for
example, calls from senior managers or top executives in an organization can be allowed to be answered first by
the destination extension.
Higher Priority can be assigned to particular Trunks, such as special or private trunk lines, trunk lines dedicated as
help lines or emergency trunks, or trunks designated as hotlines, so that when there are incoming calls on different
trunks at the same time, the call on these higher priority trunks gets answered first by the destination extension.
Priority can be assigned to all Trunk types (TWT, Mobile, SIP, T1E1PRI, BRI) and Extension types (SLT, DKP, ISDN
terminal, SIP, E&M as Station, Magneto port).
M e nu S at 0 1 0 5:3 0
Priority: 9
Fwd Busy Fwd NR Cancel Mute Conf eren c
e Tr ansfer
DN D Voi e
c M
a li Nam es Redi al Rej ect Hol d
1 2 3 de
ETERNITY
ab c f
4 gh i 5 j kl 6 mno
C LI R 7 pq r s 8 t uv 9 w xyz
C A4
H ot il ne C A3
* 0 #
S IP 2 C A2
S IP 1 C A1
Incoming Call
at 10:00:12 Mobile 1
GSM
Fwd Busy Fwd NR Cancel M ute Conf ee
rn c
e Transfer
DN D Voi c
e Ma li Names
Redi al Rej ect Hol d
1 2 ab c 3 de f
4 gh i 5 j kl 6 m no
C LI R 7 pq r s 8 tuv 9 w xyz
C A4
H ot il ne C A3 * 0 #
S IP 2 C A2
S IP 1 C A1
Here,
• There are two incoming calls, one on the Analog Trunk, TWT 1 and the Mobile Trunk 1 at the same time.
• TWT Trunk 1 has priority, ‘9’, the Mobile Trunk 1 is assigned priority ‘7’.
• Three extensions, 2001, 2002 and 3001 are calling the Operator. Extension 3001 has priority ‘7’, while
extensions 2001 and 2002 have the same priority, ‘5’.
TWT
10:00:10 9
Trunk 1
Mobile
10:00:12 7
Trunk 1
• These incoming calls, however, will appear on the Display of Operator’s phone (DKP or Extended IP
Phone) in the order of priority:
• TWT 1
• Mobile Trunk 1
• DKP 3001
• SLT 2001
• SLT 2002
• Now, when the Operator goes Off-hook (pressing speaker key or picking up the handset), the call on TWT
1 will be answered first, as TWT Trunk 1 has the highest priority.
• The Operator goes On-hook and then Off-hook, the call on Mobile Trunk 1 will be answered. Though
Mobile Trunk 1 and DKP 3001 have the same priority, ‘7’, Mobile Trunk 1 will be answered first, following
the chronological order.
• When the Operator goes On-hook after answering the call on Mobile Trunk 1, the call from DKP 3001 will
be placed on the Operator phone with a Priority Ring (configurable; default: Triple Ring).
• When the Operator goes Off-hook, the call from DKP 3001 is answered.
• When the Operator goes On-hook and then Off-hook after answering the call from DKP 3001, the call from
SLT 2001 will get answered first, though both 2001 and 2002 have the same priority, ‘5’. In this case,
Priority Ring will not be played.
• Thus, calls from trunks and extensions are answered by the landing destination in the order of priority.
Where priority is the same, calls are answered in chronological order. Calls from extensions with higher
priority are indicated by a Priority Ring on the landing destination.
Priority is relevant only when there is more than one call on the destination.
You can assign Priority to SLT extensions. However, Priority is not relevant when the SLT is a landing
destination, because there cannot be more than one call ringing on an SLT extension at a time.
To assign Priority to Trunks, you must set the priority in their “Trunk Feature Template”. See “Configuring Trunks”.
To assign Priority to Extensions, see instructions for configuring the respective Extension port type:
• “Configuring SLT Extensions”
• “Configuring DKP Extensions”
• “Configuring ISDN Terminals”
• “Configuring SIP Extensions”
• “Configuring E&M Lines”
• “Configuring Magneto Interface”
If required, you may change the Ring Pattern of the Priority Ring. See “Distinctive Rings” for instructions.
What’s this?
Extensions of ETERNITY can be protected from the intrusions by other extensions or from trunk calls by activating
Privacy.
How it works
Intrusions can occur on an extension when another extension invokes the following features:
• “DND-Override”
• “Interrupt Request (IR)”
• “Barge-In”
• “Raid”
To prevent such intrusions, ETERNITY enables you to set the following types of Privacy:
• Privacy from Interrupt Request, Barge-In, DND Override: This type of Privacy protects an extension
from intrusions by other extensions using Interrupt Request, Barge-In or DND Override.
For example: Extension A has Privacy from Interrupt Request, Barge-In and DND Override.
Extension A and B are in speech, Extension C attempts to intrude the conversation Interrupt Request or
Barge-In. Extension C’s call will be blocked and C will get error tone.
Now, Extension A has set DND and Extension B attempts to override it using DND Override. Since A has
Privacy from DND Override, B’s call will be blocked and B will get error tone.
• Privacy from Raid: This type of Privacy protects an extension from intrusions by other extensions using
Raid.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, Extension C
uses Raid to intrude the conversation. Extension C’s call will be blocked and C will get error tone.
• Privacy from Trunk call intrusion: This type of Privacy prevents the extensions in the Trunk Landing
Groups that are busy from being intruded by another waiting call. For this type of Privacy to work, the
feature “CO Call Waiting”must be disabled on the extension.
For example: Extension A is the first trunk landing extension for calls on Trunk 1. Extension A and B are in
speech. A new call lands on Trunk 1. If A has CO Call Waiting beeps disabled, A will not hear the intrusion
beeps. The system will land the call on the next extension in the trunk landing group for calls on Trunk 1.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, external caller
C uses DID to call extension A. C’s call will be blocked and C will get error tone.
How to configure
To provide Interrupt Request to extension users, you must enable this feature in the “Class of Service (COS)”
assigned to them for the time zones in their “Station Basic Feature Template”.
By default, Privacy from Raid is enabled in the Class of Service of all Extension types: SLT, DKP, SIP, ISDN. So,
none of the extensions can raid the other. You may disable this feature in the Class of Service of extensions, which
you want to protect from Raid.
By default, Privacy from Interrupt Request, Barge-In and DND Override are disabled in the Class of Service of all
Extension types. You may enable this feature on extensions which you want to protect from intrusions using any of
these features.
By default, Privacy from DID is disabled on all Extension types. You may enable this feature on extensions which
you do not want external callers to reach.
By default, CO Call Waiting is disabled on all Extension types. You may keep this feature disabled on extensions
which you want to provide Privacy from Trunk Call intrusion beeps.
For instructions, see “Class of Service (COS)”, “Station Basic Feature Template”. Also see,
• “Configuring SLT Extensions”
• “Configuring DKP Extensions”
• “Configuring ISDN Terminals”
• “Configuring SIP Extensions”
What's this?
• Q-Signaling (QSIG) is an ISDN based protocol for signaling between two PBXs. QSIG is a protocol based
on internationally agreed Standards for ISDN.
• You can network two or more ETERNITY using QSIG. This is known as 'Interoperability'.
• The basic call procedure in the QSIG is implemented as per the ECMA-143.
• The generic functional protocol for the Supplementary Services is implemented as per ECMA-165. Refer
relevant chapter for more details for the features explained in this chapter.
QSIG support is a licensed feature of ETERNITY. To use this feature you must purchase a License Key.
Refer the topic “License Management” to know more.
• Identification:
• The Calling Line Identification Presentation (CLIP)
• Calling Line Identification Restriction (CLIR)
• The Connected Line Identification Presentation (COLP)
• Connected Line Identification Restriction (COLR)
• Name Identification:
• The Calling Name Identification Presentation (CNIP)
• The Connected Name Identification Presentation (CONP)
• Call Diversion:
• Call Forward Unconditional (CFU)
• Call Forward On Busy (CFB)
• Call Forward on No Reply (CFNR)
• Call Completion:
• Call Completion on Busy Subscriber (CCBS)
• Call Completion on No Reply (CCNR)
• Recall (RE)
The implementation of these features in QSIG is as per specific ECMA standards as described below.
Advice on Charge:
• Advice on Charge (AOC) is implemented as per ECMA-211 and ECMA-212.
• Using this feature, the caller can know the cost of the call made to public N/W using networked trunk. The
cost of the call will be determined by using Call Cost Calculation. Refer chapter “Call Cost Calculation
(CCC)”
.
• The ETERNITY supports called 'AOC-E' End of the Call Charge Information. ETERNITY supports
charging for calls made to public network. The Cost of the call is calculated by the end PBX from which the
call is terminated to public Network.
CLIP/CLIR/COLP/COLR/CNIP/CONP
• The Identification features are implemented as per ECMA-163 and ECMA-164.
• Refer chapters “Calling Line Identification and Presentation (CLIP)” and “Calling Line Identity Restriction
(CLIR)”.
Call Diversion
• Call Diversion is implemented as per ECMA-173 and ECMA-174. Refer chapters “Call Forward” and “Call
Forward-Remote”.
• The CCBS is implemented as the feature Auto Call Back on Busy and the CCNR is implemented as the
feature Auto Call Back on No Reply. Refer chapter “Auto Call Back (ACB)”.
• The Call Intrusion (CI) is implemented as Raid feature. However the station on which intrusion is made, will
not get 'beeps'. On successful intrusion, the 'conference 3-party' type call will be established. See the topic
“Raid” and “Conference-3 Party”.
Call Transfer
• Call Transfer is implemented as per standards ECMA-177 and ECMA-178. Refer chapter “Call Transfer”.
• If 'CLIP-Hold' flag is disabled on the Transferring Station, the ETERNITY will send the calling station's
number as calling number. Refer chapter “Calling Line Identification and Presentation (CLIP)” and “Station
Advanced Feature Template” for more details of CLIP-Hold.
• Note 1: It is recommended to enable CLIP Hold flag for the operator station, so that when operator
transfers the call to another PBX using QSIG, the terminating PBX (ETERNITY) can identify it as call
from Public network and treat as incoming call.
• Note 2: As a Terminating PBX, the ETERNITY will consider the call as internal or from public network
depending upon the length of the digits for the calling number received.
• Note 3: If Calling number is not received because of any reason, and CLIP- Hold flag is enabled, the
ETERNITY will send the 'Trunk Name' programmed for the trunk port as calling line identification.
Recall (RE)
• Recall (RE) is implemented as per standards ECMA-213 and ECMA-214 and ECMA-143.
• The ETERNITY supports Recall-Busy and Recall-No Answer features like ETERNITY features, Call
Transfer-On Busy and Call Transfer-While Ringing.
PSTN/ISDN
Q-Sig TAC-0 GSM/VoIP etc.
DS1-1 DS1-1
2103 3103
PBX-1 PBX-2
2102 3102
2101 3101
For simple application as shown in above figure, make settings as below for PBX 1 and PBX 2.
• Now open Web Jeeves and make below settings in PBX 1 and PBX 2.
Settings at PBX 1:
Following basic settings are required in ETERNITY PBX 1.
001 21 01 0 Enable 04
002 31 01 0 Disable 04
003 0 01 0 Disable 04
• OG Trunk Bundle
Automatic Number
Trunk Port Start Total
Template Rotation Translation (ANT)
Channel Trunk
Number Port Port Type Dialed Substitute
Number Count Apply
Type No. No. List No. List
Table ID OG Ref. ID Start Channel No. Channel Count ISDN Number DDI Routing Ref. ID
01 01 01 30 2100 01
Table Reference Start DDI Total DDI DDI No. Port Port Start DDI
ID ID No. Numbers Digit Count Type Number Flexible No.
Settings at PBX 2:
Following basic settings are required in ETERNITY PBX 2.
Route Index Route Code OGTBG Strip Digit Count Self Router Flag Max. Dialed Digits
• OG Trunk Bundle
Automatic Number
Trunk Port Start Total
Template Rotation Translation (ANT)
Channel Trunk
Number Port Port Type Dialed Substitute
no. Count Apply
Type No. No. List No. List
Table ID OG Ref. ID Start Channel No. Channel Count ISDN Number DDI Routing Ref. ID
01 01 01 30 3100 01
Table Reference Start DDI Total DDI DDI No. Port Port Start DDI
ID ID No. Numbers Digit Count Type Number Flexible No.
Basic Call:
Below examples show various calls between both PBX.
Example 1:
• To make call from station of PBX 1 to any station of PBX 2 or vice versa.
• From station 2101 of PBX 1, dial station number 3101 or from station 3101 of PBX 2, dial station number
2101.
Example 2:
Example 3:
• Now External Party make IC call on Mobile port. IC call is answered by PBX 2. External Party gets DID
Music. Now External Party dials station number.
• Case 1: If External Party dials station number 3101 then call lands on 3101 of PBX 2.
• Case 2: If External Party dials station number 2101 then call lands on 2101 of PBX 1.
Supplementary Services:
Working of Supplementary Services of QSIG in ETERNITY is explained as below.
Identification:
• CLIP/CNIP:
• COLP/CONP:
When 3101 of PBX 2 answers the call then Name and Number of station 3101 is displayed on station 2101
using reverse DDI method.
• CLIR:
To restrict CLI at station 3101, below settings required for station 2101.
Here, Name and Number of station 2101 is not displayed on station 3101.
If "CLI Restriction (CLIR) Override" is enabled in Class of Service for station 3101, then Name and Number
of station 2101 is displayed on station 3101.
• COLR:
To restrict CLI when IC call answered by 3101, below settings required for station 3101.
188. Here 2101 of PBX 1 does not get dial tone after dialing 0 (TAC at PBX 2 for Trunk) because 0 is programmed in CUG table for PBX
1.
189. Refer "“c. Configuring for-IC Call using DDI Routing Over QSIG” topic for how to route Incoming Call on T1E1PRI/SIP over QSIG
using DDI Routing.
Here, Name and Number of station 3101 is not displayed on station 2101 when station 3101 answers the
IC call.
If "CLI Restriction (CLIR) Override" is enabled in Class of Service for station 2101, then Name and Number
of station 3101 is displayed on station 2101.
• Call Diversion:
Station 2101 of PBX 1 wants to set Call Diversion (Call Forward) to station number 3101 of PBX 2.
• Call Forward-Unconditional:
From station 2101 of PBX 1, dial 131-3101-#*. Now all IC calls on 2101 gets forwarded to 3101 of PBX 2.
From station 2101 of PBX 1, dial 132-3101-#*. Now all IC calls on 2101 gets forwarded to 3101 of PBX 2
when 2101 is busy.
From station 2101 of PBX 1, dial 133-3101-#*. Now all IC calls on 2101 gets forwarded to 3101 of PBX 2
when 2101 does not answer the IC call for RBT- Transfer on No Reply Timer.
• Call Completion:
Station 2101 of PBX 1 wants to set CCBS/CCNR (ACB) to station number 3101 of PBX 2.
• CCBS/CCNR:
From station 2101 of PBX 1, dial station number 3101 of PBX 2. Now dial Access code of ACB (Dial 2) to
set CCBS/CCNR on station 3101 of PBX 2.
• Call Intrusion:
Station 2101 of PBX 1 and station 3101 of PBX 2 are in speech. Now another station wants to make Call
Intrusion on station 2101 of PBX 1.
Case 1: Station 2102 of PBX 1 wants to make Call Intrusion on station 2101 of PBX 1.
• Set "Priority" level of station 2102 higher than "Priority" level of station 2101.
• Also disable "Privacy from Raid" in Class of Service for station 3101 of PBX 2.
• Now from station 2102, make call on 2101. After getting busy tone, dial Access code of Raid (Dial 5).
Speech will be established between all three parties 2101, 2102 and 3101.
If "Privacy from Raid" is enabled in Class of Service for station 2101 of PBX 1 or 3101 of PBX 2, then Raid
not possible in above case.
Case 2: Station 3102 of PBX 2 wants to make Call Intrusion on station 2101 of PBX 1.
• Disable "Privacy from Raid" in Class of Service for station 2101 of PBX 1.
• Also disable "Privacy from Raid" in Class of Service for station 3101 of PBX 2.
• Now from station 3102 of PBX 2, make call on station number 2101 of PBX 1. After getting busy tone, dial
Access code of Raid (Dial 5). Speech will established between all three parties 2101, 3101 and 3102.
If "Privacy from Raid" is enabled in Class of Service for station 2101 of PBX 1 or 3101 of PBX 2, then Raid
not possible in above case.
• Call Transfer:
Station 2101 and station 2102 of PBX 1 are in speech. Now station 2101 of PBX 1 wants to transfer the
call to any station number or external number of PBX 2.
• From station 2101, dial Flash and then dial 3101 (Station number of PBX 2) or dial 0-02652630555
(External Number).
• Now from station 2101, goes ON hook or press 'Transfer' key to transfer the call.
• User can transfer the call after making speech with second party or when second party is 'Ringing' or
'Busy'.
• Call Recall:
Station 2101 of PBX 1 transfers the call to station 3101 of PBX 2 when station 3101 of PBX 2 is 'Ringing' or
'Busy'. Now station 3101 does not answer the call.
• Call gets return to station 2101 after expiry of 'Transfer on Busy Timer' for transfer- on busy case and
'Transfer while Ringing Timer' for Transfer- while ringing case.
• Call Offer:
Station 2101 of PBX 1 and station 3101 of PBX 2 are in speech. Now station 2102 of PBX 1 wants to give
Call Offer to station 3101 of PBX 2.
• Allow "Interrupt Request" feature in Class of Service for station 2102 of PBX 1.
• Now from 2102 of PBX 1, make call on station number 3101 of PBX 2. After getting busy tone, dial
Access code of Interrupt Request (Dial 3).
• Station 3101 of PBX 2 gets beeps.
• Allow "Do Not Disturb" in Class of Service for station 2101 of PBX 1.
• Now from station 2101, activate Do Not Disturb by dialing 181.
• Now station 3101 of PBX 2 will not be able to make call on station number 2101 of PBX 1.
Station 2101 of PBX 1 has activated Do Not Disturb. Now station 3101 of PBX 2 wants to make call on
station 2101 of PBX 1.
• Allow "Do Not Disturb- Override" in Class of Service for station 3101 of PBX 2.
• Now station 3101 of PBX 2 can make call on station number 2101 of PBX 1.
If "Privacy from DND Override" is enabled in Class of Service for station 2101, then station 3101 of PBX 2
can not make call on station number 2101 of PBX 1.
Station 2101 of PBX 1 makes external call by dialing 0-02652630555 (External Number).
• After completion of call, PBX 2 gives Cost of the call using AOC.
• Cost of call in AOC is calculated at PBX 2 as per parameters set in Web pages for 'Call Cost Calculation' at
PBX 2.
• If PBX 2 does not give any charge in AOC then no charge applies on station 2101 of PBX 1.
• All calls between two stations of both PBX are free of charge.
Station 2101 and station 2102 of PBX 1 are in speech. Now station 2101 transfers the call to External Number
02652630555 using trunk of PBX 2.
• Cost of the call is given as per "Call Toggle Flag" and "Originating Flag" set in SMDR-Outgoing Calls at
PBX 2.
• If "Call Toggle Flag" is set to 'Split' at PBX 2, then charge before the call transfer is applied to station 2101
and charge after the call transfer is applied to station 2102.
• If "Originating Flag" is set to 'Originating' then all charges before and after the call transfer is applied on
station 2101 who has transferred the call.
• If "Originating Flag" is set to 'Terminating' then all charges before and after the call transfer is applied on
station 2102.
Station 2101 of PINX1 is an Operator & he wants to set Message wait on station 3101 of PINX2.
• Open "Station Advanced Feature Template" page from Jeeves of PINX2 and program "MW Notification
Type" as Stuttered Dial Tone".
• From station 2101, dial Access Code "1076 - 3101-1", to set Message wait on station 3101 of PINX2.
• Now Message wait is set on station 3101. When user goes off hook from station 3101, he will get stuttered
dial tone.
Same way Message wait can also be canceled for 3101 by dialing access code "1076-3101-0".
Configuration:
0265-304XXXX ISDN/VoIP
2199 Q-Sig
DS1-1 DS1-1
3199
PBX-1 PBX-2
2100 3100
• T1E1PRI trunk at PBX 2 has ISDN Number with MSN number 0265304 and DDI numbers are 2100-2199
and 3100-3199.
• PBX 1 and PBX 2 are connected with each other using QSIG.
• If Calling Party number is received as 0265304-2100 to 0265304-2199 then it should route on stations of
PBX 1. If Calling Party number is received as 0265304-3100 to 0265304-3199 then it should route on
stations of PBX 2.
• Settings for PBX 1 are given in Configuration of "Basic Call" and "Supplementary Services". No need to
change in these settings.
Settings at PBX 2:
• OG Trunk Bundle
Automatic Number
Trunk Port Start Total
Template Rotation Translation (ANT)
Channel Trunk
No. Port Port Type Dialed Substitute
no. Count Apply
Type No. No. List No. List
001 026530421 21
• Program Incoming (IC) Reference ID for all Time Zones as "01" in T1E1PRI Port Parameters.
Route
Route
IC Start Total DDI Route Ring on TLG Trunk
Table on TLG
Ref. Channel Channel Routing on First Timer when Feature
ID when
ID No Count Ref. ID Destination (Sec) No Templates
Busy
Reply
01 01 01 30 01 No 045 No No 01
02 01 01 30 02 No 045 No No 01
DDI No.
Table Reference Start Total DDI Port Port Start DDI
Digit
ID ID DDI No. Numbers Type Number Flexible No.
Count
• Routing Group
What’s this?
Quick Dial provides DKP and Extended IP phone users the facility of ‘One-touch’ dialing of numbers stored in their
Personal Directory and the Global Directory.
How it works
Quick Dial is based on “Abbreviated Dialing”.
• the number must exist in the Personal or Global Directory assigned to the extension.
• Personal and Global Directory dialing must be allowed in the Class of Service of the extension.
• On the DKP and Extended IP Phones, DSS keys must be configured with the Short Codes or Abbreviated
Numbers that are to be dialed out. These short codes are derived from the Index numbers of the Personal
Directory and the Memory Location Index of the Global Directory.
• You can Quick Dial a number simply by pressing the DSS key.
• The system locates the number to be dialed out in the Personal/Global Directory on the basis of the Index
Number/Memory Location Index configured on the DSS Key.
How to configure
See “Abbreviated Dialing” for instructions on configuring and assigning the Personal and Global Directories.
To assign the Short Codes or Abbreviated Numbers to be used for Quick Dial on DSS keys, for each DKP/
Extended IP Phone extension,
• List down the numbers from the Personal Directory and Global Directory to be used for Quick Dial.
• If the number is from the Personal Directory assigned to the extension, note the Index number at which it is
stored in the Personal Directory: 001 to 025.
• If the number is from the Global Directory assigned to the extension, note the Memory Location Index at
which it is stored in the Global Directory: 100 to 999.
• Now, configure the Quick Dial numbers on the DSS keys of the DKP and Extended IP Phone.
For detailed instructions on configuring DSS Keys on a Digital Key phone, see “DSS Keys Programming”.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see “Matrix Extended IP Phone
Settings” under Configuring SIP Extensions.
How to use
What’s this?
Raid allows you to interrupt a telephone conversation between two extension users, turning the conversation into a
three-way call.
You can use Raid to land in a conversation between two extension users, and between an extension user and an
external caller, with a warning beep to the extension user. The extension user will hear a beep when you raid and
you will enter in to three-way speech with both parties.
You may also Raid a conversation without any warning by disabling the beep.
How it works
• A, B and C are extension users.
• C calls A.
• If any of these three parties disconnects, two-way speech is established between the remaining parties.
Feature Interactions
• Raid works only if the dialed extension is busy in two-way speech. The two-way speech may be with
another extension or with an external number on a trunk.
• You cannot Raid on Trunks, i.e. the external number which is in two-way speech with an extension. In this
case, C can raid the conversation between A and B, but not between A and another external number.
• Raid will not work if Privacy against Raid is enabled in the Class of Service of the extension being raided.
In this case, if Extension A has Privacy against Raid in its Class of Service, C will not be able to Raid the
conversation between A and B. To know more about this feature, see “Privacy”.
• The extension using Raid must have higher Priority assigned to it than the extension being raided. In this
case, C must have higher Priority than A to be able to invoke Raid.
Raid is a sensitive feature. You are advised to restrict access to this feature to select extension users.
By default, beep is played as a warning to the extension being raided. If required, you may disable the beep played
during Raid, by clearing the Play Beep when Raid/Conference/Dial-In Conference begins check box in the
System Parameters. For instructions, see “System Parameters”.
How to use
For EON & Extended IP Phone Users
What is this?
Generally, extensions users of the PBX are given a trunk access to make outgoing calls from their phones. It is also
common for a group of extensions to share the same trunks to make outgoing calls.
When an extension user of the PBX makes an outgoing call and the called party does not answer the call or is busy
on another line, it is possible for the called party to return the call (made by the extension user) on the basis of the
CLI number received.
However, when the called party returns the call, this incoming call is mostly likely to land on the Operator extension,
as incoming calls are usually routed to the Operator.
Now, the Operator has no way of knowing which extension made the call so as to transfer the call to that extension.
Instead, the Operator must either ask the called party whom they wish to speak to and transfer the call or put the
called party on hold and find out the extension that made the call. This is an unwieldy process for all concerned -
the Operator, the called party and the extension user who originally made the call.
This can be overcome if the PBX is able to route the returned call to the original caller's extension.
ETERNITY makes this possible with the Return Call to Original Caller feature.
This feature is supported on Mobile ports, BRI, T1E1PRI (DS1) and SIP Trunks.
RCOC is not supported when calls are made from analog trunks - TWT and E&M- due to the signaling
limitations of these trunks.
This feature requires a license. This feature will work if you have the license for the Mobility Feature Suite
or the Business Feature Suite. For more information, see “License Management”.
How it works
The Prerequisites
• RCOC is enabled on the desired Trunk/s - BRI, T1E1PRI, Mobile and SIP.
• RCOC is enabled in the Class of Service group assigned to the extension.
The Process
• When an extension having RCOC feature in its Class of Service makes an out going call, the system
checks if RCOC is enabled on the trunk through which the outgoing call is routed.
• If RCOC is enabled on the trunk, the system stores the record of the outgoing call in an internal database
referred to as the RCOC Table.
• The system sets RCOC for the outgoing call in the following conditions, according to the Destination
Port190:
• If the Destination Port is a BRI or T1E1PRI (DS1) Port or a SIP Trunk, RCOC is set when:
• If the Originating Port is either BRI-NT or T1E1-NT, the system checks the Class of Service allowed
to the trunk port.
• If the Originating Port is a trunk (T1E1-TE, BRI-TE, TWT, MOBILE, SIP), RCOC will be set, if it is
enabled on Destination Port
• RCOC shall be set only if the Calling Party's Number is available. If calling party number is missing,
then RCOC shall not be set.
• Whenever there is an incoming call on any trunk, the system matches the CLI of the incoming all with the
RCOC Table.
• If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
• The return call rings on the original caller's extension for the period of the Ring Back Tone Timer
(programmable; default 45 seconds). If the original caller does not answer the call within this Timer, the call
is routed to the Trunk Landing Group programmed for that trunk.
• If no match is found in the RCOC Table or the extension or the original caller is busy, the call will be routed
according to the incoming call logic programmed (as programmed in the assigned Trunk Feature
Template) in the system.
• When DISA CLI Authentication (Multiple Calls or One Call) is enabled on a trunk, whenever there is an
incoming call on the trunk, the system will first check the DISA CLI Authentication Table.
• If a matching entry is found in the DISA CLI Authentication table, the system will give dial tone to the caller.
• The caller can now invoke RCOC feature by dialing ** (pressing Star key twice).
OR
• The caller can make calls to a station or an external number or use a feature as required.
• If the caller invokes RCOC feature by dialing ** (pressing Star key twice), the system will check the RCOC
Table.
As RCOC is a “Class of Service (COS)” based feature, extensions that are not allowed this feature in their
COS cannot have their calls returned; even if this feature is enabled on the Trunk they used to make the
call.
• The record of the outgoing call is stored in the RCOC Table, only if the same record does not already exist
in the database.
• Each entry is kept for the duration of the RCOC Record Delete Timer (programmable; default: 999
minutes). Whenever a record is stored in the RCOC database, the Record Delete Timer for that entry is
activated. On the expiry of the Timer, the entry is deleted by the system.
• Each record is deleted from the database either after the call is returned or on expiry of the Record Delete
Timer.
• In case of Call Transfer, RCOC will be set for the extension which made the call on the trunk.
• The Ring Back Tone Timer is common to all internal calls; calls made from one extension will ring on the
destination extension till the end of this timer. Change in the Ring Back Tone Timer for RCOC returned
calls on original caller's extension will also be applied on Ring Back Tone Timer for all internal calls. So,
program this Timer taking this into consideration.
• Persons using DISA must be informed about RCOC feature access code ** and how to use this feature
when in DISA mode.
How to configure
For this feature to work, it must be enabled on the Trunk and in the Class of Service of the stations. If desired, the
related Timers, i.e., the RCOC Record Delete Timer and the Ring Back Tone Timer may also be changed.
RCOC on Trunk
• Click the trunk parameters of the trunk type on which you want to enable this feature, namely:
• SIP Parameters
• BRI Parameters
• Exit SE mode.
In the default Station Basic Feature Template 01 assigned to all stations of the ETERNITY, the default Class of
Service group 01 has the feature "RCOC" enabled. So, all stations of ETERNITY are by default allowed this
feature.
There is no need to program this feature if all stations are to be allowed this feature.
However, if you want to deny this COS feature to certain stations and allow this feature to all other stations, follow
these steps:
Refer the topic “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
What’s this?
Various features and facilities supported by ETERNITY, such as Alarms, Station Message Detail Records, System
Activity Log, Time Zones, Daylight Savings, certain Voice mail features need the correct time and date for their
proper functioning.
The ETERNITY has a built-in Real Time Clock (RTC) circuit that maintains date and time. When you select Region,
the RTC is automatically set to the current date and time of the country/region where ETERNITY is installed.
Since the RTC circuit may drift over a period, it is recommended that you check and reset RTC values at least once
every month to correct this drift. The RTC of ETERNITY takes care of leap years.
How to configure
First set the date format and then set the date.
• Exit SE mode.
Time Zones
041 Egypt(GMT+02:00)
042 Fiji(GMT+12:00)
043 Finland(GMT+02:00)
044 France(GMT+01:00)
045 Germany(GMT+01:00)
046 Greece(GMT+02:00)
047 Guyana(GMT-04:00)
049 Hungary(GMT+02:00)
050 India(GMT+05:30)
051 Indonesia(GMT+07:00)
052 Iran(GMT+03:30)
053 Iraq(GMT+03:00)
054 Ireland(GMT)
056 Italy(GMT+01:00)
057 Japan(GMT+09:00)
058 Jordan(GMT+02:00)
059 Kazakhstan(GMT+06:00)
060 Kenya(GMT+03:00)
063 Kuwait(GMT+03:00)
064 Kyrgyzstan(GMT+06:00)
065 Lebanon(GMT+02:00)
066 Libya(GMT+02:00)
067 Malaysia(GMT+08:00)
068 Maldives(GMT+05:00)
069 Mauritius(GMT+04:00)
073 Mongolia(GMT+08:00)
074 Mozambique(GMT+02:00)
075 Myanmar(GMT+06:30)
076 Namibia(GMT+01:00)
077 Nepal(GMT+05:45)
078 Netherlands(GMT+01:00)
080 Nigeria(GMT+01:00)
081 Norway(GMT+01:00)
082 Oman(GMT+04:00)
083 Pakistan(GMT+05:00)
084 Paraguay(GMT-04:00)
085 Peru(GMT-05:00)
086 Philippines(GMT+08:00)
087 Poland(GMT+01:00)
088 Portugal(GMT)
089 Qatar(GMT+03:00)
090 Romania(GMT+02:00)
094 Singapore(GMT+08:00)
095 Slovakia(GMT+01:00)
097 Spain(GMT+01:00)
099 Sudan(GMT+03:00)
100 Sweden(GMT+01:00)
101 Switzerland(GMT+01:00)
102 Syria(GMT+02:00)
103 Taiwan(GMT+08:00)
104 Tajikistan(GMT+05:00)
105 Thailand(GMT+07:00)
106 Turkey(GMT+02:00)
107 Uganda(GMT+03:00)
108 Ukraine(GMT+02:00)
111 United States (Atlanta, Augusta, Boston, Charlotte, Columbus, Detroit, Indiapolis,
Miami, NY, Philadelphia, Washington)(GMT-05:00)
112 United States (Chicago, Dallas, Des Moines, Memphis, Minneapolis, New Orleans,
Oklahoma, Omaha, St. Louis) (GMT-06:00)
113 United States (Albuquerque, Boise, Cheyenne, Denver, Salt Lake City) (GMT-07:00)
114 United States (Las Vegas, Los Angeles, Phoenix, San Francisco, Seattle) (GMT-
08:00)
117 Uzbekistan(GMT+05:00)
118 Venezuela(GMT-04:30)
119 Vietnam(GMT+07:00)
120 Yemen(GMT+03:00)
121 Yugoslavia(GMT+02:00)
122 Zambia(GMT+02:00)
123 Zimbabwe(GMT+02:00)
What's this?
Reminders are a variation of the “Alarms” feature, requiring the Date and Time to be set for each Reminder call.
Reminder calls are useful for extension users who wish to be reminded of important tasks or appointments.
For Reminder calls, date and time are set in the following format:
Date is set, according to Date Format you selected in the “Real Time Clock (RTC)” parameters, as:
• Day-Month-Year (DD:MM:YYYY)
Or
• Month-Date-Year (MM:DD:YYYY).
• Multiple Reminders can be set for an extension by the Operator and/or by the extension user.
• The mechanism for serving Reminders calls can be configured as 'Personalized' or 'Automated'.
• Reminders can be voice-guided, if the ETERNITY has a Voice Mail System (VMS) installed in it.
• ETERNITY can register as many as 48 Reminders set by the Operator and extension users.
How it works
Personalized Reminder
When the Reminder call serving mechanism is configured as 'Personalized',
• The Operator Phone rings first191, displaying the number of the extension to which the reminder call is to
be served.
• When the Operator answers this call, a call is placed on the extension on which the reminder call is set.
• The extension phone rings for the duration of the Alarm Ring Timer.
• When the extension user answers the call, the Operator greets the extension user with the reminder
message.
191. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the ETERNITY will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.
• If the extension is busy192, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
• inform the extension user about the Reminder in person or send someone to do it.
OR
OR
Personal Reminders will work even if the extension user has set DND or Call Forward.
Automated Reminder
When the Alarm serving mechanism is configured as 'Automated',
• The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is a
DKP or the Matrix Extended IP Phone, Reminder message will appear on its display.
• When the extension user answers the call, the user may be played music-on-hold, or a pre-recorded voice
message, or be connected to the Voice Mail, or routed to the Operator, depending upon the Alarm
Notification Type you have configured for the extension.
• If the extension user does not answer the reminder call, the ETERNITY makes two more attempts (in all, 3
attempts) at an interval of 5 minutes between each attempt, to call the extension.
• If all Reminder call attempts go unanswered, the ETERNITY places the call on the Operator Phone. The
Operator Phone rings till the end of the Alarm Ring Timer. The Operator Phone displays the number of the
extension with the message 'No Reply'. The Reminder call is now considered as served.
• If the extension phone is busy, the ETERNITY will continue to make the Reminder call Attempts at the
Alarm Interval programmed. When all Alarm Attempts go unanswered, ETERNITY will place a call on the
Operator phone. The Operator Phone will display the number of the extension phone with the message
'Busy'.
192. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'.
• The extension phone rings for the Number of Alarm Attempt configured, at the set Alarm Attempt
Interval.
• The extension stops ringing when the user answers the call and dials 0 to acknowledge the Reminder
call. This reminder call Acknowledgement Code 0 is non-configurable.
• Reminder settings will be retained in the system during power down and system upgrades.
• When multiple reminder requests have been set by an extension user, the extension user cannot
selectively cancel a particular reminder request. Only the Operator can selectively cancel Reminders
set for an extension user from the System Administrator pages of Jeeves.
• It is not possible to modify—change the date and time—of a reminder request. So, you may cancel the
Reminder request and set a new one.
The status of Reminders set by Operator as well as extension users appears on this page, with details of time
(hours and minutes), and serving mechanism (personalized, automated).
The Operator can view the Reminder report whenever required and can also print this report.
How to configure
The configuration of Reminders is the same as Alarms.
• Configure, as required, the Alarm Call related parameters: Alarm Ring Timer, Number of Attempts,
Alarm Attempt Interval, Configurable Alarm Type and Configurable Alarm Category, and Snooze.
• Configure Macros, if the SLT extension has special function keys, and you want to a function key for the
Reminder feature.
• Select Destination Port for Reminder Reports: COM Port or Ethernet Port.
If the Voice Mail System (VMS) is installed in the ETERNITY, it can offer voice-guided Reminders to extension
users and the Operator.
Voice-guided reminders lead users through a menu, helping them set the alarm in a step-by-step manner.
For SLT
If the SLT of the extension user has a special Reminder function key, the extension user can set the alarm using
this key.
• Press 'Reminders' key. (The label on the SLT key may differ from model to model)
• Follow the Voice Mail System prompts to set/cancel reminders.
• Without the Voice Mail System installed, the extension user having SLT with the special Reminder
function key will not be able to set/cancel Reminders. This extension user can set/cancel Reminders
only by dialing the feature access code for voice-guided Reminders.
OR
• Lift handset.
• Dial 1072-033
• Dial Extension Number.
• Dial Date and Time in the format:
DDMMYYYYHHMM
OR
• MMDDYYYYHHMM (users in USA)
• Dial 1 for Personalized, Dial 2 for Automated.
• You get confirmation tone.
• Replace handset.
• Lift handset.
• Dial 1072-033
• Dial Extension Number.
• Dial #
• You get confirmation tone.
• Replace handset.
To cancel reminder calls selectively, go to 'Reminder Status' page from the System Administrator of
Jeeves.
To set Reminder:
To cancel Reminder:
• Press 'Reminder' Key again.
OR
• Dial 162
• Select 'Cancel All'.
• Press Enter Key.
To set Reminder,
• Lift handset.
• Dial 162
• Dial Date and Time in the format
DDMMYYYYHHMM
OR
MMDDYYYYHHMM (users in USA)
• You get confirmation tone.
• Replace handset.
• Open Jeeves.
• Select the Cancel Reminder check box of the extension number for which you want to cancel the
reminder.
• Click the Cancel Selected Reminders button at the bottom of the page.
• If the date format of ETERNITY is set as MM-DD-YYYY or the Region 'USA' is selected, then the
reminder report will be printed according to this date format. To know more, see “Real Time Clock
(RTC)”.
• Dial 1072-038.
• Dial 1072-039
• Dial the scheduled time in HHMM format (from 00:00 to 23:59 hours; default: 00:00).
For example, dial 1730 to set the time to 17 hours and 30 minutes.
• Replace handset.
The report will be printed at the scheduled time on the destination port assigned for printing Hotel/Motel reports.
What’s this?
• One can program ETERNITY from any remote location. Direct inward system Access (DISA) facility of the
system allows a remote user to login and use most of the functions of the system. Programming is one of
such functions allowed to the remote user. The remote user can program the system using the same
commands as used by the normal local station to program the system. The user can login into the SA
programming or SE programming mode.
How to use
• Make a DISA call.
• Login as DISA user. Get the DISA login beeps. Enter the SE/SA mode. Get programming/Dial Tone.
• In case none of the trunk lines have DISA facility and it is required to do remote programming, then follow
following steps:
SE Mode
Dial 1#91-SE Password program the system. Dial ‘00’ exit from the programming mode (SE Mode).
SA Mode
Dial 1#92-SA Password program the system. Dial 1#92 exit from SA mode.
How it works
Following flow chart depicts the process:
Is the No
password
correct ?
Error Tone
Yes
Is the
Yes command dialed No
correct and accepted
by the system ?
Confirmation tone Error tone
Exit
Program
Mode Wait for error
Wait for confirmation
tone to get over
tone to get over System gives dial tone
End
Relevant Topic:
1. “Direct Inward System Access (DISA)” 1156
What’s this?
This feature enables the DKP and Extended IP Phone Extension users to listen to the conversations taking place in
another location where a DKP/Extended IP Phone is present.
Room Monitor can be used to monitor activities on the Shop Floors / Manufacturing areas from another location.
How it works
• A is a supervisor in a Manufacturing unit.
• A’s room in on the second floor. The manufacturing area is on the ground floor.
• To keep track of the activities in the plant on the ground floor, there must be a DKP or an Extended IP
Phone at the place where the activities are to be monitored, and A’s extension must have higher “Priority”
than the extension at the monitored location.
• If there is a DKP or an Extended IP Phone at the desired location, A can activate Room Monitor.
A can activate Room Monitor only if the DKP/Extended IP Phone at the desired location is idle.
• When A activates Room Monitor, the microphone of the DKP/Extended IP Phone on the ground floor goes
Off-hook. A can now hear all the sounds taking place on the ground floor, without anyone present there
coming to know that they are being monitored.
• Room monitoring will be terminated on the DKP/Extended IP Phone on the ground floor, if someone lifts
the handset of this phone or if there is a call on this phone from another extension.
• You can activate Room Monitor from any extension port type, but the extension being monitored must
be a DKP or an Extended IP Phone.
How to configure
To be able to use Room Monitor, extension users must have this feature enabled in the “Class of Service (COS)” in
the “Station Basic Feature Template” assigned to their extensions.
What’s this?
A Routing Group is a group of extensions which can function Many times it is required to call any one of the
persons among a group of related people. It is not important to talk to a particular person. The caller just wants any
member from the group. ETERNITY offers flexibility to put related stations in a group. These groups are called
Routing Groups. The call made to this group using the Access Code of the Routing Group. However the caller can
access individual stations of the group also by dialing the station number. The routing group is used to land IC call
on any trunk port.
Refer chapter “System Parameters” for command to put the station as 'in service' or 'out of service'.
How it works
• Maximum 96 Routing groups can be formed.
• The station can be a SLT, DKP, DOP, OG Trunk Bundle Group, SIP Extension or ISDN Terminal.
• The sequence in which various stations in the group should ring can be arranged.
• Once a station receives a ring, it can be set to ring continuously till the call matures. Such a station
continues ringing even when other stations of the group are hunted. This is called Continuous ringing and
can be programmed for each station.
• If the call is not answered even after hunting the last station, the system will loop back and start from the
first station once again.
• A fresh call can start hunting either from the first station or from the final station of the previous call. This
method is called Rotation Method and can be set for each group. If rotation method is enabled, the fresh
call will land on the destination next to the one, which received the last call. This would enable equal
distribution of incoming calls to all the destinations within the group. If the rotation method is disabled, the
fresh call will always land on first station of the department group. Refer chapter “Outgoing Trunk Bundle”.
• User can program the 'OG Trunk Bundle Group' in the Routing Group. Hence, the ANT feature will be
automatically applicable as OG Trunk Bundle is part of OG Trunk Bundle Group.
• Now, the trunk to trunk calls will have advantage of LCR feature as LCR is property of OG Trunk Bundle
Group.
• Other Parameters Ring Timer, Continuous Ring and Rotation will remain fixed.
• Now the system will identify the access code (Flexible number) associated with the Software port of ISDN
terminal programmed in the Routing group and will include the flexible number in Called party number
field.
• The CLI received on the Trunk port will be forwarded to ISDN Terminal in "Calling Party Number" field.
How to configure
The commands explained below should be referred as:
• To program a single port: XXXX-1
• To program a range of ports: XXXX-2
• To program all the ports: XXXX-*
Step 1
Use following command to Program the destination in the Routing Group:
6502-1-Routing Group-Number Index-Port Type-Port Number
6502-2-Routing Group-Routing Group-Number Index-Port Type-Port Number
6502-*-Number Index-Port Type-Port Number
Where,
Routing Group is from 01 to 96.
Destination Index is from 01 to 32.
00 Nonea 000
01 SLT 001-512
02 DKP 001-128
10 DOP 001
Step 2
Use following command to program the time for which each station in the group should ring:
6503-1-Routing Group-Member Index-Ring Timer
6503-2-Routing Group-Routing Group-Member Index-Ring Timer
6503-*-Member Index-Ring Timer
Where,
Routing Group is from 01 to 96.
Member Index is from 01 to 32.
Ring Timer is from 001 to 255.
Step 3
Use following command to program continuous or non-continuous ring for a station in the group:
6504-1-Routing Group-Member Index-Flag
6504-2-Routing Group-Routing Group-Member Index-Flag
6504-*-Member Index-Flag
Where,
Routing Group is from 01 to 96.
Member Index is from 01 to 32.
Flag Meaning
Step 4
Use following command to program rotation method of a department group:
6505-1-Routing Group-Rotation Method
6505-2-Routing Group-Routing Group-Rotation Method
6505-*-Rotation Method
Where,
Routing Group is from 01 to 96.
0 Fresh call lands on the first station within the group (disable continuous)
On issuing the command all routing groups will get defaulted to the following values.
Member Index Port Type Port Number Ring Timer (Sec.) Continuous Ring Rotation
Relevant Topics:
“Department Call”
“Voice Message Applications”
“Trunk Landing Group (TLG)”
“Security Alarm and Reporting”
What's this?
With the Security Alarm and Reporting feature of ETERNITY, external numbers as well as extensions can be
notified about emergency situations such as fire, break-in, burglary, etc.
This is a useful security feature. For example, in the event of burglary in a Bank, the teller or any employee can
press the Emergency Alarm193. The system will dial out the programmed external numbers such as Police,
Security Agency or to another branch office number to report the emergency, without anyone coming to know about
it.
Security Alarm
Security Alarm makes use of the “Digital Input Port (DIP)” to function.
A panic switch, a smoke detector or a break-in detector can activate the DIP. The system will sense the event and
will place a call to the numbers programmed to receive security alarm calls. These numbers may be
The system will play a pre-recorded voice message to inform the called parties (external numbers or group of
extensions) of the emergency or activate the siren connected on the DOP as programmed.
It is possible to select a different destination for Security Alarm calls according to Time Zones, i.e. working hours,
break hours, and non-working hours.
For example, factory manager can have the extension of Security Personnel programmed for Security Alarm during
working hours and break hours, and can have an external number, such as the factory manager’s residence phone
number, programmed for Security Alarm for non-working hours.
Security Reporting
It is possible to program the system to 'report' Security Alarm calls made to external numbers simultaneously also
to a group of extensions. Hence, the feature name Security Reporting.
The system will display the DIP Port number with the Text message 'Emergency' to the extensions on which it
lands the call. The extension user can know the location of the emergency if s/he knows the location of the sensor/
panic switch connected to the DIP Port Number displayed on his/her phone.
How it works
The Pre-requisites
• An emergency switch/sensor, output of a Smoke detector, Glass break detector, Fire Alarm, etc., is
connected as instigator to the “Digital Input Port (DIP)” of ETERNITY.
• A “Time Tables” defining the working hours, break and non-working hours is assigned as the Time Table
for Security Alarm.
• For each Time Zone, the destination for Security Alarm calls is selected as DOP or External Numbers, or
Routing Group and the corresponding DOP/External Number/Routing Group of extensions is assigned.
• The Security Reporting flag is enabled in the Security Alarm Parameters, to report the emergency with the
prerecorded voice message to a group of extensions.
• Security Alarm - Delay Response Timer: the time of initiating the dialing after the Security Alarm is
triggered by the DIP.
• Call Attempt Interval for External Number: the time gap between each attempt to call an external
number.
• Number of Attempts for each External Number: the number of attempts the system should make to
dial each external number you have programmed (in case no acknowledgement is received from the
number).
The Process
• The sensor instigates the DIP.
• The system waits for the duration of the 'Minimum Instigation Time' you programmed (default: 01 sec) to
respond to the instigation.
• On expiry of this timer, the system waits for the Security Alarm - Delay Response Timer (programmable;
default: 15 seconds), while checking the 'Trigger Security Alarm on' destination programmed for the
current Time Zone.
• At the end of the Delay Response Timer, it places a call to the first external number.
When Security Reporting flag is enabled it also simultaneously places a call to the first extension number
in the Security Reporting-Routing Group.
194.Enable the port, set the Instigation Signal and the Minimum Instigation Time as required.
195.The pre-recorded message provided by Matrix (on the Product CD) is: "This is an emergency call. Please dial '0' to acknowledge".
Refer the topic “Voice Message Applications”.
• When the external called party answers the call, the ETERNITY delivers a pre-recorded Emergency
message (recorded in the Voice Module).
• The system repeats the voice message until the external and the internal called parties dial '0' to
acknowledge the call.
If the internal called party number is as a digital key phone, the system will display the text message for
Security Alarm Acknowledgement.
• When there is no response from the first external number, the ETERNITY dials the second external
number programmed for the current Time Zone.
If there is no response from the second external number or if the number is busy, the system tries the third
external number programmed for the current Time Zone and tries to deliver the message.
Thus, the system tries each number, one after the other, as many times as programmed in the Number of
Attempts (default: 5 attempts) with a time interval programmed in the Call Attempt Interval (default: 15
seconds).
The system stops dialing each external number only when the called party acknowledges the call by
dialing '0' or when the Number of Attempts is over.
When Security Reporting flag is enabled, the system also simultaneously places a call to the first
extension number in the Security Reporting-Routing Group.
If the Reporting group extension is a digital key phone, the system will display the text message <<DIP-1
Emergency>> on the phone display.
• When any of the Security Alarm-Routing Group extensions answers the call, the ETERNITY delivers the
pre-recorded Security Alarm message (recorded in the Voice Module).
It repeats the voice message until the called extension dials '0' to acknowledge the call.
• If there is no response from the first Security Reporting-Routing Group extension number, the system will
follow the Routing Group Logic to try other extensions in the group to land the Security Alarm call.
The system stops attempting the Security Alarm call, when any of the extensions in the Security Alarm-
Routing Group answers the call and acknowledges the call by dialing '0'.
If the Security Alarm-Routing Group extension is a digital key phone, the system will display the text
message for Security Alarm Acknowledgement.
• The Security Alarm 'Number of Attempts' count and the 'Call Attempt Interval' are not applicable for
routing Security Alarm calls to internal numbers (Routing Group).
How to configure
For this feature to work, you must program the Security Alarm parameters and the Digital Input Port.
Before you set the Security Alarm Parameters, ensure that you have the following programmed and ready:
• The sensor device connected to the Digital Input Port and the port parameters programmed. Refer the
topic “Digital Input Port (DIP)” for instructions.
• A voice message containing an Emergency Message is recorded in the Voice Module for Security and
Emergency. Refer the topic “Voice Message Applications” for instructions on recording voice module.
• Routing Groups of stations, i.e. Security Alarm-Routing Group, on which Security Alarm should be
triggered, if you have selected as 'Routing Group' for any of the Time Zones.
• A Routing Group of stations, i.e. the Security Reporting-Routing Group, on which security reporting should
initiated.
• The Time Table for Security Alarm, defining the working hours, non-working hours and break hours for the
Time Zones of the Response Type selected. Refer the topic “Time Tables” for programming instructions.
Now, program the Security Alarm Parameters using Jeeves or by dialing SE commands from a Telephone.
• Use DIP to trigger Security Alarm: Enable this flag to use the Security Alarm feature. If this flag is
disabled, no alarm will be triggered. By default the flag is disabled.
You may program the Digital Input Port, if not done already. To do this,
Whether you are using the DIP for an automated control application or for Security Dialing and Reporting, you must
program the following parameters:
• Time Table for Security Alarm: Select the Time Table you want to apply for Security Alarm. By default,
Time Table 1 is selected.
• You may program the time table, if not done already. To do this,
Click the link 'Time Table' in this parameter.
• The Time Table page will open.
• Configure the default Time Table, or another Time Table.
• Click 'Submit' at the bottom of the page to save changes.
• Return to the Security Alarm page.
• Select the Time Table you programmed for this parameter.
Any changes you make in the Time Table will affect the features/applications which use the same Time
Table. Program Time Tables taking this into account. Program different Time Tables for features/
applications so that they remain unaffected.
• Trigger Security Alarm during Working Hours on: Select the desired destination - DOP, External
Number, Routing Group - on which Security Alarm should be initiated during the Time Zone - Working
Hours. Depending on the destination you selected, program the following parameters:
• DOP number: If you selected DOP, select the DOP number. ETERNITY PE supports 3 DOPs,
whereas ETERNITY ME and GE support only a single DOP.
• Routing Group number: If you selected Routing Group as destination, select the number of the
Routing Group you programmed for Security Alarm. For example, if you have programmed Routing
Group number 03 for Security Alarm for Working Hours, select 03. By default, Routing Group 01 is
selected.
• External Number 1: If you selected External Number as the destination, enter the first external number
you want the system to dial for Security Alarm.
• External Number 2: Enter the second external number you want the system to dial for Security Alarm.
196.'High' state signifies that the DIP is normally open. DIP should be programmed as 'High' when the sensor connected to the DIP
keeps the Loop open and closes it to signal an event.
197.'Low' state signifies that the DIP is normally closed. DIP should be programmed as 'Low' when the sensor connected to the DIP nor-
mally keeps the Loop closed and opens/breaks it to signal an event.
198.This is the time for which the instigation signal from the sensor device should remain present on the DIP to be recognized by the DIP
as a genuine signal.
The External Numbers may be numbers of Emergency Services like the Police, Ambulance, or of any
key decision makers.
• Trigger Security Alarm during Break Hours on: Like the previous parameter, select the desired
destination - DOP, External Number, Routing Group - on which Security Alarm should be initiated during
the Time Zone - Break Hours. Depending on the destination you selected, program the DOP Number/
Routing Group/External Numbers, as described above.
• Trigger Security Alarm during Non-Working Hours on: Repeat the instructions given above to select
the desired destination for Security Alarm during the Time Zone - Non-Working Hours and program the
DOP Number/Routing Group/External Numbers.
• Security Reporting: Select the check box to enable this flag, if you want Security Reporting to be
initiated for Security Alarm calls. Security Reporting enables notification of emergency situations to
Routing Group member extensions.
• Security Reporting-Routing Group: If you enabled Security Reporting, select the number of the
Routing Group you programmed for Security Reporting. For example, if you have programmed Routing
Group number 04 for Security Reporting, select 04. By default, Routing Group 01 is selected.
• Security Alarm - Delay Response Timer (sec): enter the time for which the system should wait
before dialing out the first external number after receiving signal about the emergency from the DIP. By
default the Delay Response Timer is set to 15 seconds.
• Number of Attempts for each External Number: If you programmed External Numbers as Security
Alarm destination for any of the three Time Zones, you may program this parameter. This parameter
defines the number of attempts the system should make to dial each external number you have
programmed. When no acknowledgement is received from an external number you programmed, the
system will repeatedly call the number for the number of Attempts you programmed. By default, the
number of attempts is set to 5.
• OG Trunk Bundle Group to Dial External Number: If you programmed External Numbers as
Security Alarm destination for any of the three Time Zones, select the Outgoing Trunk Bundle Group
from which the system should dial the external number. By default, OG trunk bundle group 01 is
selected.
• If you have finished programming the Security Alarm parameters, click 'Submit' to save your settings.
To select the Security Alarm 'Trigger on' destination for a Time Zone, dial:
• 5204-TimeZone-Trigger on
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
Trigger on destination is
1 for DOP
2 for Routing Group
3 for External Number.
To program Routing Group for Security Alarm for a Time Zone, dial:
• 5206-TimeZone-Routing Group
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
Routing Group is from 01 to 96.
By default, Routing Group 01 is selected for all Time Zones.
To program External Numbers for Security Alarm for a Time Zone, dial:
• 5207-TimeZone-Index-External Number-#*
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
Index is from 1 to 3.
External Number is a number string of a maximum 16 digits. Terminate the command with #* if the
number of digits in the External number is fewer than 16.
• Exit SE mode.
How to use
This is an automatic application. The system automatically dials the programmed numbers on receiving the signal
from the DIP. Human intervention is required only for acknowledging the emergency call placed by Security Alarm.
It is unlikely that external called parties would know that '0' must be pressed to acknowledge the
emergency call. You are recommended to include this information in the Voice Module you record for
Security Alarm prompting the called party to dial '0'.
It may happen that Security Alarm is initiated mistakenly. In which case, it must be terminated from the SA mode.
What’s this?
ETERNITY supports different extension and trunk port types. In the Selective Port Access feature, each port type is
assigned a Port Access Code. Extension users can access a particular port by dialing the Port Access Code
assigned to the Port and its Port Number.
How it works
• Extension user A wants to access a particular Mobile port, Mobile Port 1 to make a call. Extension A must
dial the Selective Port Access Feature Code, followed by the Port Type Code for Mobile ports and then dial
the Port Number.
• By default, the following access codes are assigned to each Port Type:
04 BRI 01 to 32
05 T1E1 1 to 8
25 Mobile 01 to 64
26 SIP Trunk 01 to 32
28 ISDN Terminal 01 to 64
Here, Extension A must dial 69-25-01, where 69 is the feature code for Selective Trunk Access, 25 is the
port access code for the Mobile Port, and 01 is the number of the Mobile Port which A wants to access.
Similarly, if Extension A wants to access SIP Extension 10, A must dial 69-34-010.
How to configure
To be able to use Selective Port Access, extension users must have this feature enabled in their “Class of Service
(COS)”.
What’s this?
You can use Self Ring Test to check the functioning of your own extension phone. Self Ring Test allows you to call
your own extension. You can check the ringing volume of your extension phone.
How to use
What's this?
The ETERNITY supports Balance Inquiry and Recharging of the SIM Card installed in its Mobile Ports199.
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic “License Management” to know more.
How to use
To be able to use this feature, first collect the following information from your Network Operator:
• Balance Inquiry Number: This is the number provided by the Network Operator to the subscribers to
check Balance. Different Network Operators have different numbers. For example, the Balance Inquiry
number of Vodafone is *141#.
• Recharging Service Number: This is the number provided by the Network Operators to their subscribers
for Recharging Service. Different Network Operators have different numbers for Recharging Service. For
example, the Recharging Service Number of Vodafone is *140*.
199. ETERNITY supports Unstructured Supplementary Service Data (USSD), the standard for transmitting information over CSM sig-
naling channels and a commonly used method to query the available balance and other similar information in pre-paid GSM ser-
vices.
• All the mobile ports configured in the system will appear on this page, by their Names you programmed
when configuring the mobile trunk ports.
If you have not programmed any name for a port, the Name field for that port will appear blank.
Balance Inquiry
• To make Balance Inquiry,
• Click the "Request' button of Balance Inquiry, for all those Mobile Ports for which you want to request
Balance Inquiry.
• Enter the Balance Inquiry Number provided by the Network Operator whose SIM Card you have installed
in the Mobile Port.
A maximum of 8 digits are allowed. The valid digits for Balance Inquiry number are any digits from 0 to 9
and the characters * and #
• Click 'Submit'.
• Enter the following information in the appropriate fields under 'Recharge' for the Mobile Ports:
• Number: Enter the Recharging Service Number provided by the Network Operator in this field.
A maximum of 8 digits are allowed. The valid digits for Recharging Service number are any digits from
0 to 9 and the characters * and #
• PIN: Enter the PIN number which is printed on the Recharge Voucher/Coupon. Your Recharge PIN
number may consists of a maximum of 20 digits.
The valid digits for Balance Inquiry number are any digits from 0 to 9 and the characters * and #.
Make sure you enter the digits and characters of the Recharge PIN number exactly as given on the
Recharge Voucher/Coupon.
• For each port that you send a Balance Inquiry/Recharge Request, you will get this USSD-Reply: "Please
wait, processing the request. Refresh the page to see the current status."
• The response received from the GSM network (including possible error messages) will be displayed under
'USSD-Reply'. When the USSD Reply is received from the network, it will appear with the Date and Time
stamp of ETERNITY in this field.
• For each Mobile Port (SIM Card) at a time you can either request Balance Inquiry or Recharge the SIM
Card. When you select the radio button of Balance Inquiry of Mobile Port 1, 'Recharge' will be disabled
for Mobile Port 1. And when you select the radio button of Recharge, 'Balance Inquiry' will be disabled
for Mobile Port 1.
However, you can send Balance Inquiry/Recharge request for all the Mobile Ports available in the
system.
• During Balance Inquiry/Recharge-Request, the status of the Mobile port will be 'busy'. It will become
idle only after the USSD response is received from the GSM network.
• The ETERNITY will clear the USSD Reply after system restart. So each time you open the 'SIM
Balance and Recharge' page after system restart, the USSD Reply box will be blank.
What’s this?
The ETERNITY supports different types of ports as illustrated below:
Port
The ETERNITY treats a port as an entity and processes it on the basis of port type and its programmed attributes.
Type of Port
SLT
DKP
Trunk
DOP
DIP etc.
Attributes of Port
Software
Port Access Code
COS Group
Toll Control Group
DID, etc.
Hardware Port
Slot Number
Port Number
Software Port
The ETERNITY takes a software port as fundamental entity. It processes the software port. Hardware ID and the
access code are just two attributes of a software port and hence they are not used anywhere in processing and
programming.
BRI 32 01-32
T1E1PRI 8 1-8
Each type of software port has different attributes. The System Engineer (SE) programs these attributes using the
corresponding template at the time of installing the ETERNITY.
Hardware ID is an attribute of a software port. Hence, all the programming is done for the software port and not for
the hardware ID. Accordingly, the software port number is used for all programming tasks.
An access code is just a Flexible number assigned to the software port. Programming is for the software port and
not for the access code (Flexible number). Accordingly, the software port number is used for all the programming.
The System Engineer allocates software port numbers to different users. This allocation is Flexible and any
software port number can be assumed for any user. Hardware ID is not relevant at this stage. Hardware ID can be
programmed for a software port any time. Further, it can be changed any time in case of hardware failure of a port.
• Software port numbers start from 001 for all different port types.
Hardware ID
The hardware ID of a software port denotes where the port is physically located. To derive hardware ID of a
software port, we need:
To clear the hardware ID assigned to a SLT software port use the following command:
1101-SLT-00-00
Use the following command to clear the hardware ID assigned to DKP software port:
1102-DKP-00-00
Use the following command to clear the hardware ID assigned to the DSS software port:
1103-DKP-DSS-00-00
Use the following command to clear the hardware ID assigned to TWT software port:
1104-TWT-00-00
Use the following command to clear the hardware ID assigned to E&M software port:
1105-E&M-00-00
Use following command to de-assign the hardware slot and the hardware port assigned to the BRI software port.
1106-BRI-00-00
Use the following command to clear the hardware ID assigned to T1E1PRI software port:
1107-T1E1-00-00
Use following command to de-assign the hardware slot and the hardware port assigned to the Mobile port.
1108-Mobile Trunk Number-00-00
Use following command to de-assign the hardware slot and the hardware port assigned to the Magneto port.
1110-Magneto port-00-00
Relevant Topics:
1. “Paging” 1366
2. “Music on Hold (MOH)” 1335
3. “DSS Keys Programming” 482
What's this?
Static Routing Table is required when you have more than one router (gateway) in your network and you want
ETERNITY to send packets to multiple routers/gateways for different types of calls.
Static Routing Table helps route calls between point to point sites (connected through Multi Protocol Label
Switching-MPLS, Frame Relay, etc.) and to public internet at the same time.
How it works
For example, two Local Area Networks, Network A and Network B, are connected through Frame Relay/ Multi
Protocol Label Switching (MPLS) network to give access to local resources and also to make Peer-to-Peer calls.
59.162.252.82
SIP Proxy
A B
192.168.1.0/24 192.168.2.0/24
Public
IP
Frame Relay/MPLS
192.168.1.1
ETERNITY ETERNITY
The Static Routing Table makes it possible to route different types of outgoing calls—Peer to Peer or Proxy—made
to different subnets through different Gateways.
The Static Routing Table defines the appropriate Gateway Address (or Router’s LAN Address) where the IP
packets are to be sent.
When ETERNITY sends packets, if the final destination IP Address and ETERNITY are not in the same Subnet, the
system will check the Static Routing Table.
If a perfect match is found, ETERNITY will start sending the IP packets to the corresponding Gateway Address
configured in the table.
If no match is found, ETERNITY will send the IP Packets to the Default Gateway Address (Network Connection
Type) you configured in the Master Ethernet Port Parameters. See “Configuring Master Ethernet Port Parameters”.
How to configure
The Static Routing Table must be configured at each location where ETERNITY is installed. You may configure the
Static Routing Table using Jeeves or by dialing the system commands from a telephone connected to the
ETERNITY.
The Static Routing Table allows you to configure up to 8 entries. Each entry is stored against an Index
number.
• Destination Address: This is the address of the final destination where the call is to be made. This
can be a device IP Address or Network Address.
• Gateway Address: This is the IP address of the node where the IP packets are to be sent. Generally,
it is the IP address of the LAN interface of the Router.
• The Gateway Address 192.168.1.1 specifies the LAN address of the Router A which connects location
A and location B.
The IP address of the LAN interface of the router which connects Location A to the public internet
should be configured as Default Gateway in the Network Parameters of ETERNITY in location A.
With the Static Routing Table configured thus, all calls made by ETERNITY to 192.168.2.0/ 24 will be
routed through the router which connects Location A to Location B. Whereas, all calls made by
ETERNITY to addresses other than 192.168.2.0/ 24 will be routed through the Default Gateway.
Similarly, configure the Static Routing Table in ETERNITY at location B to enable calling from Location
B to Location A.
• 7811-1-Index-Destination Address
Where,
Index is 1 to 8.
Destination Address is of 15 digits maximum. Enter each octet in full. For example, to program
192.168.10.10, dial 192168010010.
• Exit SE mode.
What’s this?
• The ETERNITY can record the details of Internal, Incoming (IC) and Outgoing (OG) calls made from/to all
the stations. This is called Station Message Detail Recording (SMDR).
• The SMDR can be obtained as a report, only if storage of each type of calls is enabled. To get the SMDR
user will be required to set the filter conditions and assign the destination port. The online SMDR is also
supported by the ETERNITY which gives the report immediately after the call is made or received.
• The ETERNITY supports third party call cost calculation protocols and the parameters for this can also be
set by configuring the SMDR. Thus the user will be required to program following types of parameters:
• SMDR Storage: These parameters are programmed to enable the storing of the IC, OG and Internal
calls. Refer chapter “Station Message Detail Recording-Storage”.
• SMDR Report: These parameters are programmed to assign destination port for getting report of IC,
OG and Internal calls and to get offline report. Refer chapter “Station Message Detail Recording-
Report”.
• SMDR Online: These parameters are programmed to assign destination port for getting online report
of IC, OG and Internal calls and to configure the call record format for IC calls. To interface third party
call accounting software (CAS) with ETERNITY, SMDR online mode must be enabled. Refer chapter
“Station Message Detail Recording-Online”.
• SMDR Posting: ETERNITY supports call cost calculation and call record formats as Holidex, Hobic,
Hobis, Xiox, etc. Using call cost calculation, ETERNITY calculates cost of the call and sends the call
record to the PMS (Property Management Software) using protocols, like Holidex, Hobic, Hobis, etc.
The parameters for this are programmed to select the protocol for call accounting software and to
configure parameters for OG Handshaking Protocol and OG call record format. To assign destination
port and destination IP Address for OG posting parameters. Refer chapter “Station Message Detail
Recording-Posting”.
Refer separate Manual for Hotel Applications for more details about PMS.
Relevant Topics:
“Station Message Detail Recording-Storage” 1545
“Station Message Detail Recording-Report” 1531
“Station Message Detail Recording-Online” 1473
“Station Message Detail Recording-Posting” 1489
What's this?
• The ETERNITY can generate report for the calls as and when the call is made provided the printer or a
computer is connected to the respective port. This is called SMDR Online report.
• To get the online report SE should assign the destination port for IC, Internal and OG calls. SE should
ensure that the storage flag is enabled to store the calls in the buffer of the system. Refer chapter “Station
Message Detail Recording-Storage” to enable the storage of calls.
• The ETERNITY supports to change the default format for the SMDR Report , such as column position and
field length for calling number, speech duration, remarks field, etc. SE can configure these parameters as
per user's requirement.
For example,
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
How to configure
Internal Calls: (Assigning destination port and using start/stop command for Online report)
Use following command to assign a destination port for Online SMDR-Internal call record:
2830-Code
Where,
Code Meaning
0 None
1 COM1
2 COM2
3 Printer Port
4 Ethernet Port
By default, the port assigned is None. This means the Online printing is disabled.
If you assigned Ethernet Port as destination port, use the following command to assign the IP Address to the
Ethernet Port:
2832-IP Address
By default, IP Address is 192.168.1.104
• On assigning the communication port or the printer port or Ethernet Port as the destination port, the report
generation is directed to that as soon as the incoming call is completed.
• SE should take care, not to overlap the assignment of same ports to the different processes.
Flag Meaning
0 Abort
1 Start
By default, Flag is 0.
• The ETERNITY provides a facility to abort the report generation in midway (1072-136-0). Once the report
generation is aborted, then it has to be explicitly started with (1072-136-1). This command is issued from
the SA mode.
• OG Online Report (Assigning destination port and using start/stop command for Online report)
Use following command to assign destination port for Online SMDR-OG Call Record:
2730-Code
Where
Code Meaning
0 None
1 COM1
2 COM2
3 Printer
4 Ethernet Port
By default, the port assigned is None. This means the On-line printing is disabled.
If you assigned Ethernet Port as destination port, use the following command to assign the IP Address to the
Ethernet Port:
2732-IP Address
By default, IP Address is 192.168.1.104
• On assigning the communication port or the printer port or Ethernet Port as the destination port, the report
generation is directed to the assigned port as and when an OG call is completed.
Flag Meaning
0 Abort
1 Start
By default, Flag is 0.
The ETERNITY provides a facility to abort the report generation in midway (1072-101-0). Once the report
generation is aborted, then it has to be explicitly started with command (1072-101-1). This command is issued from
the SA mode.
IC Online Report (Assigning destination port and using start/stop command for Online report)
Use following command to assign destination port for Online SMDR-IC Call Record:
2930-Code
Where,
Code Meaning
0 None
1 COM1
2 COM2
3 Printer Port
4 Ethernet Port
By default, the port assigned is None. This means the Online printing is disabled.
• On assigning the communication port or the printer port or Ethernet Port as the destination port, the report
generation is directed to that as soon as the incoming call is completed.
• SE should take care, not to overlap the assignment of same ports to the different processes.
Flag Meaning
0 Abort
1 Start
By default, Flag is 0.
The ETERNITY provides a facility to abort the report generation midway (1072-151-0). Once the report generation
is aborted, then it has to be explicitly started with (1072-151-1). This command is issued from the SA mode.
Serial Number
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
Reset Meaning
1 No Compulsory Reset
Increment Counter
Reset Meaning
1 No Compulsory Reset
Property Code
Use following command to program property code string for property code:
8209-Property Code String
• This code is required by the Property Management System (PMS) when it is catering to more than one
PMS interfaces.
• Refer separate Manual for Hotel/Motel Applications for more details about PMS and Hotel applications
for this feature.
Station Number
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
Trunk Number
1 Matrix Format
2 Check-In Format
• First Character in Check In Format is X (Fixed). Remaining three characters show the software port
number. However, this will not specify whether the call is made through TWT 125 or E&M 125. Also the
channel number will not be specified in case of call made through T1E1PRI port or BRI port.
Date
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
01 DD-MM-YY
02 DD/MM/YY
03 DD.MM.YY
04 DD MM YY
05 DDMMYY
06 DD-MM-YYYY
07 DD/MM/YYYY
08 DD.MM.YYYY
09 DD MM YYYY
10 DDMMYYYY
11 MM-DD-YY
12 MM/DD/YY
13 MM.DD.YY
14 MM DD YY
15 MMDDYY
16 YY-MM-DD
17 YY/MM/DD
18 YY.MM.DD
19 YY MM DD
20 YYMMDD
21 YYYY-MM-DD
22 YYYY/MM/DD
23 YYYY.MM.DD
24 YYYY MM DD
25 YYYYMMDD
26 MM-DD
27 MM/DD
28 MM.DD
29 MM DD
30 MMDD
31 DD-MM
32 DD/MM
33 DD.MM
34 DD MM
35 DDMM
Use following command to program date fill flag for date field:
8257-Date Fill Flag
Where,
0 Disable
1 Enable
By default, Date Fill Flag is ‘1’ (i.e. single digit in Date, Month and year is printed with prefix ‘0’).
• Leading Zeros field is applicable for Date, Month and Year i.e. whether the single digit date is to be
printed as space-X or 0-X. For example: date = 1 is to be displayed as ‘1’ or ‘01’. In case when leading
zeroes are not required, the date, month and year sub-fields are right aligned and the spaces are filled
with character ‘space’.
• This Date field is not linked to the global flag of Date Format. The global Flag of Date format is used
while using features or in configuration reports but not in PMS. This is because the date format used by
the PMS is not the same as used by the users of the system.
Time
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
0 Disable
1 Enable
Use following command to program time fill flag for time field:
8258-Time Fill Flag
Where,
0 Disable
1 Enable
Answer Duration
Use following command to program column position for answer duration field:
8227-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 54.
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
Use following command to program fill character for answer duration field:
8230-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is ‘Space’.
Use following command to enable/disable the Filler character flag for Answer Duration:
8259-Filler Character Flag for Answer Duration
Where,
0 Disable
1 Enable
Use following command to program duration unit for answer duration field:
8231-Duration Unit
Where,
1 HH:MM:SS
2 HHMMSS
3 Minutes
4 Seconds
When Duration Unit = Minutes, rounding to nearest whole number is done. For seconds <= 30, Minute is
not incremented and for seconds > 30, minute is incremented.
Use following command to program column position for hold duration field:
8232-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 58.
Use following command to program field length for hold duration field:
8233-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 03.
Alignment Meaning
1 Left Alignment
2 Right Alignment
Use following command to program fill character for hold duration field:
8235-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is ‘Space’.
Use following command to enable/disable the Filler character flag for Hold Duration:
8260-Filler Character Flag for Hold Duration
Where,
0 Disable
1 Enable
Speech Duration
Use following command to program column position for speech duration field:
8237-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 62.
Use following command to program field length for speech duration field:
8238-Field Length
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
Use following command to program fill character for speech duration field:
8240-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is ‘Space’.
Use following command to enable/disable the Filler character flag for Speech Duration:
8261-Filler Character Flag for Speech Duration
Where,
0 Disable
1 Enable
Called Number
Use following command to program column position for called number field:
8242-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 00. (This field is not available by default)
Use following command to program field length for called number field:
8243-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 16.
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 1.
Use following command to program number format for called number field:
8245-Number Format
Where,
1 Continuous
2 Separated
When separated is selected, put ‘-‘ in the called party number of 10 digits. First after three digits and
another after six digits.
Calling Number
Use following command to program column position for calling number field:
8246-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 06.
Use following command to program field length for calling number field:
8247-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 16.
Use following command to program alignment for calling number field:
8248-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 1.
Use following command to program number format for calling number field:
8249-Number Format
1 Continuous
2 Separated
When separated is selected, put ‘-‘ in the called party number of 10 digits. First after three digits and
another after six digits.
DID Digits
Use following command to program column position for DID digits field:
8250-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 00. i.e. this field is not available by default.
Use following command to program field length for DID digits field:
8251-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 00.
Alignment Meaning
1 Left Alignment
2 Right Alignment
Remarks
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 1.
Relevant Topics:
1. “Station Message Detail Recording” 1472
2. “Station Message Detail Recording-Storage” 1545
3. “Station Message Detail Recording-Report” 1531
4. “Station Message Detail Recording-Posting” 1489
The Station Message Detail Record (SMDR)-Posting feature of ETERNITY is used for interfacing the system with
CAS.
SMDR-Posting sends call detail records to CAS for the purpose of call cost calculation.
When ETERNITY is interfaced with a third party Call Accounting Software (CAS) (See Note1) to determine the cost
of the calls made by the station users, the system uses SMDR-Posting to send to CAS call record details, like
number to which the call was made by the station user, number of the station from which the call was made, the
date and time when the call was made, the duration of the call, metering pulses incurred for the call, etc. On receipt
of this information, the CAS calculates the cost of the call for billing.
As different CAS interfaces support different protocols, the ETERNITY offers the flexibility to send call detail
records using the protocol supported by CAS. SMDR-Posting supports as many as 15 different widely-used CAS
protocols such as, Holidex, Hobic, Micros A, Micros B, Comm One, Call-Inn, Bell-HOBIC, XIOX, RSI and others.
Each posting protocol has its own handshaking protocol and call record format. The Installer/System Engineer can
configure any one of these depending upon the protocol supported by CAS. It is also possible to customize the
posting protocol to match the settings required by the CAS used by the organization.
SMDR-Posting is supported on Serial RS232 Communication Port as well as on TCP/IP Ethernet Port. Thus, the
CAS can be interfaced on either the COM port or the Ethernet port of the ETERNITY. For every outgoing call, call
detail record is posed on the designated port (COM port/Ethernet port)
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic “License Management” to know more.
SMDR-Posting Protocols
The ETERNITY supports as many as 15 different posting protocols from the system to CAS. The flow of messages
between the ETERNITY and various protocols of CAS Interface is described in the following.
1. Holidex/HOBIS A
This protocol is used by Amstar, CLS, Compass, Compu-solve, Dehan, Encore, Fabco, HIS, Holidex, HRGAS,
InnSolutions, Inn-Star, Lodgemate, Logistix, Omron, Otto Clerk, Reserve 5, Resort Computer, RDP, Springer-
Miller Systems, Star and Stuart.
HOBIS B is used by EECO and New Systems Protocol for transfer of messages from the PBX to CAS.
ENQ >
< ACK
STX-(tex)-ETX-BCC >
< ACK
ENQ >
The PBX will retransmit an ENQ after 5 seconds until the CAS accepts the message or until 4 NAK responses
are received.
No Response
The PBX will retransmit an ENQ after 5 seconds until CAS responds or until 4 Unsuccessful ENQ responses
have been sent.
ENQ >
< ACK
STX-(tex)-ETX-BCC >
< NAK
< ACK
The PBX will make a maximum of 3 attempts to send the message. If the message is still not transmitted
successfully, it will drop the message and proceed to the transmission of the next message.
ENQ >
< ACK
No Response
ENQ >
< ACK
Control Character
Meaning
Char. ASCII Value
ACK Positive Acknowledgement by the CAS (Indicates successful reception of data by HEX 06
the CAS)
NAK Negative Acknowledgement by the CAS (Unsuccessful reception of data by the HEX 15
CAS)
STX This marks the beginning of the data transfer. It also starts the accumulation of the HEX 02
BCC
ETX This is the last data character. Marks the end of the data. It is immediately followed HEX 03
by the BCC.
BCC This is a 'block check character' used to verify the successful transfer of data Depends on
between the systems. BCC is calculated by processing through an accumulatory the data
by an Exclusive OR operation. The BCC process should start with the character
after the STX character.
2. HOBIS B
Handshaking Parameters for HOBIS B are as below:
<LF> -message-<CR>
ACK
<LF>-message-<CR>
NAK
<LF>-message-<CR>
NAK
<LF>-message-<CR>
NAK
<LF>-message-<CR>
The ETERNITY will make 3 tries (default) to send the message for Data Transfer Retry Count - on Negative
Response. If the ACK is still not received from the CAS, the ETERNITY will proceed to the next message.
(no response)
After sending the message the ETERNITY will wait for the response to data timeout time. If no response is
received from CAS for the sent message, the ETERNITY will log this message in the System Fault Log and
look for new message to be sent to CAS.
3. Hobic
Protocol for transfer of messages from PBX to CAS.
SOM-(tex)-EOM >
< ACK
< NAK
< NAK
The PBX will make a maximum of 3 attempts to send the message. If the message is still not transmitted
successfully, it will drop the message and proceed to the transmission of the next message.
No Response
The PBX will wait for 5 seconds to receive response from the CAS. If no response is received from the CAS
during this time period, it will drop the message and proceed to the transmission of the next message.
SOM This marks the beginning of the message. Start of Message. HEX 0A (Line Feed)
4. Micros A
Handshaking protocol for Micros A shall be as shown below:
PBX CAS
<y
The first field of Text is always 'ac01' which marks the start of text. However, this field is a part of message and
not a link control character.
PBX CAS
<n
The PBX will make a maximum of 2 attempts to send the message. If the message is still not transmitted
successfully, it will drop the message and proceed for the transmission of next message.
PBX CAS
No Response
The PBX will wait for 5 seconds and if no response is received from the CAS, it will drop the message and
proceed to the transmission of the next message.
5. Micros B
Handshaking protocol for Micros B is as shown below:
PBX CAS
< ACK
The first field of Text is always 'ac01' which marks the start of text. However, this field is a part of message and
not a link control character.
PBX CAS
< NAK
PBX CAS
No Response
The PBX will wait for 5 seconds and if no response is received from the CAS, it will drop the message and
proceed to transmit the next message.
7. RSI-CMS
ETERNITY sends call detail record in following format to RSI-CMS call accounting interface.
STX-Message-EXT
Call accounting interface will not send any response, ACK/NAK, for the messages received.
HOLIDEX:
Start
Field Filler/
Parameter Column Alignment Remarks
Length Character
Number
Currency 33 01 NA NA $
Symbol
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
• If the actual dialed number is less than the specified field width, the number will be sent as per the
programmed alignment.
• If the dialed number length is greater than the width of the dialed number field, the trailing number digits
will be removed. For example: if the number dialed is 15134036508 (11 digits) and the width of the
called number field is 8, then the first 8 digits will be sent.
HOBIS A:
Increment 04 01 NA NA --
Counter
Currency Symbol 33 01 NA NA $
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
HOBIS B:
Same as HOBIS A, except the handshaking parameters.
HOBIC:
Currency Symbol 33 01 NA NA $
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
By default the link control character EOM will be Carriage Return (CR - HEX 0D), Line Feed (LF - HEX 0A)
and Form Feed (FF - HEX 0C)
BELL HOBIC:
Currency Symbol 33 01 NA NA $
No Area Code is
provided.
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Remarks 00 NA NA NA --
MICROS A:
Start Column
Parameter Field Length Alignment Fill Char. Remarks
Number
Serial Number 00 NA NA NA --
Increment Counter 00 NA NA NA --
Property Code 00 NA NA NA --
Trunk Number 00 NA NA NA --
Date 00 NA NA NA --
Time 00 NA NA NA --
Duration 00 NA NA NA --
Units 00 NA NA NA --
Currency Symbol 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Remarks 00 NA NA NA --
MICROS B:
Same as MICROS A, except the handshaking parameters.
HILTON:
Currency Symbol 29 01 NA NA $
Location 00 NA NA NA --
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
In this protocol, the ETX is at the column position 61; hence the PBX will send blanks from column position 50 to
60.
XIOX:
Currency Symbol 32 01 NA NA $
Location 00 NA NA NA --
Units 00 NA NA NA --
Trunk Number 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
Comm One:
Serial Number 00 00 X X X X
Increment
00 01 X X X X
Counter
Property Code 00 00 X X X X
Units 00 00 X X X X
Amount 00 00 X X X X
Currency 00 00 X X X X
Call Type
00 00 X X X X
Indicator
Location 00 00 X X X X
Remarks 00 00 X X X X
Call-Inn:
Serial Number 00 00 X X X X
Increment
00 01 X X X X
Counter
Property Code 00 00 X X X X
Trunk Number 00 NA X X X X
Units 00 00 X X X X
Amount 00 00 X X X X
Currency 00 00 X X X X
Call Type
00 00 X X X X
Indicator
Location 00 00 X X X X
Account Code 00 00 X X X X
Remarks 00 00 X X X X
RSI-CMS:
Increment Counter 00 00 X X X X
Property Code 00 00 X X X X
DD/MM/
Date 38 10 RA Yes 032
YYYY
Units 00 00 X X X X
Amount 00 00 X X X X
Currency 00 00 X X X X
Location 00 00 X X X X
Remarks 00 00 X X X X
When SMDR-Posting Protocol is selected as 'Customized', then the various parameters of the Call Detail
Record format can also be customized.
When the Call Detail Record format is customized, if there is a gap between two fields, these fields will be
'space' (ASCII-32).
ETERNITY supports CAS Interface on Communication Port (RS232) as well as Ethernet port (TCP/IP).
Depending upon the installation scenario of the ETERNITY in the organisation, the Installer/System Engineer
may decide whether to use the CAS interface on the COM Port or on Ethernet Port of the system.
If the Installer/System Engineer has decided to set up the CAS Interface on the COM Port (Refer Note2) (RS232),
the following functional components are required to make the interface work:
Now, locate a spare serial/COM port on the PC. Connect the COM port the ETERNITY with the COM port of
the PC using the communication cable supplied by Matrix (Refer Note3).
If the Installer/System Engineer has decided to set up the CAS Interface on the Ethernet Port (TCP/IP), the
following functional components are required to make the interface work:
• A PC with a spare Ethernet port (not supplied by Matrix) Or any free Ethernet Port of the LAN Switch on
which the CAS server application software is running.
Now, connect the Ethernet port of the Master/CPU card of the ETERNITY with the Ethernet Port of the PC (on
which CAS server application is running) or to one of the Ethernet ports of the LAN Switch, if the CAS server
is in the same LAN.
• Enabling storage of Outgoing (OG) SMDR. By default, OG SMDR storage is enabled. Refer “Station
Message Detail Recording-Storage”.
• If SMDR-Posting is through TCP/IP (i.e. the CAS Interface is to be set up on the Ethernet port),
program the destination IP address and Port.
• To disable dial 2701-0. If this flag is disabled, the system will not store records of outgoing calls.
Destination Port:
Code Meaning
0 None
1 COM1
2 COM2
3 Not Used
4 Ethernet
01 Blind Send
02 Matrix
03 Holidex
04 HOBIS A
05 HOBIS B
06 HOBIC
07 BELL HOBIC
08 MICROS A
09 MICROS B
10 Hilton
11 Xiox
12 Comm One
13 Call-Inn
14 RSI-CMS
15 Customized
When the above command is issued, the system will set the default values of the handshaking related parameters
and call record format parameters of the selected Posting Protocol.
The following table shows the default values for Handshaking parameters and Link Control parameters of various
OG-SMDR Posting protocols:
Parameter/
Type 1 2 3 4 5 6 7 8 9 10 a b c D e f g
Protocol
01 Blind Send 1 1 1 1 1 1 1 1 1 1 0 000 000 000 000 CR-LF 0
02 Matrix 3 5 3 5 3 3 5 3 5 3 0 000 ACK NAK STX ETX 1
03 Holidex 5 4 5 4 5 11 3 11 3 11 1 ENQ ACK NAK STX ETX 1
04 HOBIS A 3 3 3 3 3 3 3 3 3 3 1 ENQ ACK NAK STX ETX 1
05 HOBIS B 3 3 3 3 3 3 3 3 3 3 0 ENQ ACK NAK LF CR 0
06 HOBIC 5 3 3 3 3 5 3 3 3 3 0 000 ACK NAK LF CR-LF- 0
FF
07 BELL 5 5 5 5 5 5 3 3 3 3 1 ENQ ACK NAK STX ETX 1
HOBIC
08 MICROS A 5 3 3 3 3 5 3 3 3 3 0 000 y N 000 CR 0
09 MICROS B 5 3 3 3 3 5 3 3 3 3 0 000 ACK NAK 000 CR 0
10 HILTON 3 3 3 3 3 3 3 3 3 3 0 000 ACK NAK STX ETX 1
11 XIOX 2 3 2 3 2 2 3 2 3 2 0 000 ACK NAK STX ETX 1
12 Comm One 1 1 1 1 1 1 1 1 1 1 0 000 000 000 000 CR-LF 0
13 Call-Inn 1 1 1 1 1 1 1 1 1 1 0 000 000 000 000 000 0
14 RSI-CMS 1 1 1 1 1 1 1 1 1 1 0 000 000 000 STX ETX 0
15 Customized 1 1 1 1 1 1 1 1 1 1 0 000 000 000 000 CR-LF 0
Code Meaning
0 abort
1 Start
• ENQ Retry Count - on No Response (to ENQ): The number of times the sender should send ENQ before
dropping the process, in case response is not received for the last message sent.
• ENQ Retry Count - on Negative Response (to ENQ): The number of times the sender should send ENQ
before dropping the process, in case of a negative response received for the last message sent.
• ENQ Retry Time - on Negative Response (to ENQ): The time after which the sender should sent the
ENQ again.
• Response to Data Timeout: The time for which the sender waits for a response to data from the receiver.
• Data Transfer Retry Count - on No Response (to Data Transfer): The number of times the sender
should send ENQ before dropping the process. This parameter is used when ACK is received against
ENQ and there is some problem while sending the data.
• Data Transfer Retry Time - on No Response (to Data Transfer): The time after which the sender should
send the ENQ again before dropping the process. This parameter is used when ACK is received against
ENQ and there is some problem in sending the data.
• Data Transfer Retry Count - on Negative Response (to Data Transfer): The number of times the
sender should send ENQ before dropping the process. This parameter is used when ACK is received
against ENQ and there is some problem in sending the data.
• Data Transfer Retry Time - on Negative Response (to Data Transfer): The time after which the sender
should sent the ENQ again before dropping the process. This parameter is used when ACK is received
against ENQ and there is some problem in sending the data.
• Use ENQ Character: This flag is to be enabled if the protocol uses ENQUIRE (ENQ) Signal.
• ENQ Character: The ASCII character (Single Character) used to send ENQUIRE (ENQ) signal to the
receiver.
• Acknowledgement (ACK) Character: The ASCII character (Single Character) used by the receiver to
acknowledge the receipt of the Link Control Character/Message Data.
• No Acknowledgement (NAK) Character: This parameter signifies the ASCII character (Single
Character) used by the receiver to dis-acknowledge the receipt of the Link Control Character/Message
Data.
• Start of Packet Character: A string of four ASCII characters used by the receiver to indicate Start of
Packet. Each ASCII character is from 000 to 252. Start of Packet may be of one character only, in which
case the string should be completed by programming remaining three characters with ASCII Null
Character (000).
• End of Packet Character: A string of four ASCII characters used by the receiver to indicate End of
Packet. Each ASCII character is from 000 to 252. End of Packet may be of one character only, in which
case, the string should be completed by programming the remaining three characters should be
programmed as ASCII Null (000).
• Use Byte Code Check (BCC): This flag is to be enabled when the protocol uses BCC Signal.
• Enter SE mode.
0 Disable
1 Enable
ENQ Character
0 Disable
1 Enable
This may be required if a 'customized' protocol has been selected by the Installer/System Engineer.
The Call Detail Format for OG-SMDR Posting Protocols consists of the following parameters. For each parameter
explained briefly below, you can define the column position, field length (i.e. the number of digits), the alignment
(whether left aligned or right), and the filler characters, wherever required. Refine the following format parameters
according to the type of posting protocol you have selected and the requirement of the CAS being used by the
organization.
Serial Number: This is the serial number generated for each call record. Serial numbers are generated from
000 to 999. When serial number '999' is reached, the numbers roll over to 000.
Increment Counter: It increments when the serial number counter rolls over. The Increment counter starts
from A, ending at Z, and then roll over back to A.
Property Code: This is the property code required by the CAS used in the organisation. It is a string of
alphanumeric characters and is to be terminated with #*. This field has a maximum of 128 alphanumeric
characters.
• The System Engineer must program this string keeping in mind the field length used by the selected/
customized posting protocol.
• The default value of the default Property Code String has been set as 'AAA', as at least two known
protocols use this field. The System Engineer can set a different value here and the new value will
appear in the CDR record, irrespective of the protocol type selected.
• If Bell Hobic or Hilton has been selected, the System Engineer should program this field as 'AAA'. If
Xiox protocol has been selected, the System Engineer should program this field as HTL. These values
are not protocol dependent, but can be configured by the System Engineer.
Station Number: This is the extension number from which the call was made. The System Engineer can
define the column position and the field length of the Station number in the Call Detail Record.
Trunk Number: This is the number of the trunk from which the call was made.
• The First Character in the Check-Inn Format is X (Fixed). The remaining three characters show the
software port number. However, this does not specify whether the call is made through TWT 125 or
E&M 125. Also, the channel number is not specified in case of call made through T1E1PRI port or BRI
port.
Date: The date on which the call was made. The date fill flag is to be enabled.
• Filler Character field is applicable for Date, Month and Year, i.e. whether the single digit date is to be
printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
• Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
spaces are filled with character 'space'.
• The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not in CAS. This is because the date format used by
the CAS is not the same as used by the users of the system.
Time: The time when the call was made. The format of the time field and the time fill flag are to be
programmed.
• Filler Character field is applicable for Hours, Minutes and Seconds i.e. whether the single digit hour is
to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
Duration: The duration of each call. Program the duration unit and the duration fill flag.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
Units: The duration of the call interpreted in terms of units. The number of units depends on the Pulse Rate.
The number of units is derived from the Call Unit = Call duration in seconds/Pulse rate in seconds.
Amount: This is the Amount of the call. Program the amount format and the fill flag.
• Filler Character field is applicable for both the sub fields of Amount viz. Rupees/Paisa i.e. whether the
single digit Rupee is to be printed as space-X or 0-X. For example, Rupee = 1 is to be displayed as '1'
or '01'. Where leading zeroes are not required, the Rupee and Paisa are right aligned and the spaces
are filled with character 'space'.
• When Amount Format = Higher Currency, rounding to nearest whole number is done. For Lower
Currency <= 50, Higher Currency is not incremented and for Lower currency > 50, Higher Currency is
incremented.
Currency: This is the symbol of the currency in which the Amount is charged. A maximum of 8 ASCII
Characters are allowed.
• Generally, Currency Symbol field prefixes to Amount field. Hence, to comply with various CDR formats,
it is recommended that the column position of Currency Symbol and Amount field should be
programmed properly.
• The System Engineer can change the Currency Symbol used in the OG-SMDR Format. However, this
change will not be reflected in the Front Desk User Wizard.
Call Type Indicator: This indicates the type of call made, i.e. whether local, international, information, etc.
The System Engineer must program the Number String, the Text String and its Meaning as explained in
following table:
01 0 LD Long Distance
02 95 IC Inter Circle
04 0 INTL International
: : : :
36 2 L Local
The Number Index is kept as '36' as one of the SMDR-OG Posting protocols, INN-FORM XL supports 24
different types of calls.
By default, all the entries in this table are blank.
The System Engineer is advised to program the first 10 entries of this table as below if the selected posting
protocol is Bell Hobic or XIOX.
01 1 A
02 2 A
03 3 A
04 4 A
05 5 A
06 6 A
07 7 A
08 8 A
09 9 A
10 0 A
: : :
The System Engineer is advised to program the first 11 entries of this table as below, if the selected
posting protocol is Holidex or Hobic.
01 1 L
02 2 L
03 3 L
04 4 L
05 5 L
06 6 L
07 7 L
08 8 L
09 9 L
10 0 L
11 0 F International
: : :
• The Text String should preferably be same as Field Length. If not, the remaining spaces will be filled
with character 'Space'. If the Field length is less than the Text string characters, then the number of text
characters equal to the Field length will be printed.
Location: This column indicates the location of the external number to which the call was made.
• The system detects the location from the called location programmed in the Area and Country Code
Tables.
• Called Location is programmed as one of the parameters of the Area Code Table and Country Code
Table. Depending upon the prefix dialed, the Location string is picked up from either Country Code
table or Area Code table.
• The Called Location parameter in the Country Code table and Area Code table is of 8 Characters.
• If the number of characters in the field Called Location is more than Field length then the remaining
characters will not be printed (overlapped by next field).
• If the number of characters in the field Called Location is less than Field length then the remaining
characters in the field Called Location will be filled by spaces.
Called Number: This is the external number to which the call was made.
• One way to separate the called party number is by Area Code, Exchange code and Subscriber
Number. This is difficult in an Open numbering system, in which the field size of area code, exchange
code are not standard but vary from two digits to four digits (for example, the Area code for 'Mumbai' is
of 2 digits, whereas that of 'Vadodara' is 3 digits).
• In the Closed numbering system, the Area Code, Exchange Code and the Subscriber number are of
fixed length. In such case, including '-' in the called party number is not difficult. Hence, '-' is put in the
called party number. The called party number is assumed to be of 10 digits. The first '-' is placed after
four digits, counting from the right. The second '-'is placed after seven digits, counting from the right. If
the dialed number is a local number of 7 digits then the second '-'is not placed. Also, the remaining
three digits are not placed, but filled with character 'space'.
• In this case, even if the call is made to a geographical area where open numbering system is followed,
'-' is placed in the same way.
Account Code: This is the Account Code (Refer Note4) using which the call was made.
Remarks: This column indicates the details of the call; whether it was a DISA call, DOSA call, Auto Redial
Call, type of call maturity.
Fixed Characters are used to indicate the type of call, call details, etc. The notations for the Remarks field are:
D DISA Call
K 12KHz/16KHz
R Reversal
D Delay
I Connect
Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial number
counter will not be automatically reset.
Starting Character - Increment Counter: Specify the starting character of the increment counter as the
serial number rolls over, in this field.
Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00 HH:MM) or
every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial number counter will not
be automatically reset.
Prefix String Required: This flag is to be programmed if the prefix string 0ac1 is to be sent when interfacing
with OG-SMDR Posting Protocol.
• Enter SE mode.
Serial Number
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
Reset Meaning
1 No Compulsory Reset
Increment Counter
Reset Meaning
1 No Compulsory Reset
Use the following command to program starting character for increment counter:
8174-Starting Character
Where,
Starting Character is from A to Z.
Property Code
Use following command to program property code string for property code:
8109-Property Code String
Where,
Property Code String is maximum of 16 characters.
By default, it is 'AAA'.
Station Number
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
1 Matrix Format
2 Check-Inn Format
Date
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
01 DD-MM-YY
02 DD/MM/YY
03 DD.MM.YY
04 DD MM YY
05 DDMMYY
06 DD-MM-YYYY
07 DD/MM/YYYY
08 DD.MM.YYYY
09 DD MM YYYY
10 DDMMYYYY
11 MM-DD-YY
12 MM/DD/YY
13 MM.DD.YY
14 MM DD YY
15 MMDDYY
16 YY-MM-DD
17 YY/MM/DD
18 YY.MM.DD
19 YY MM DD
20 YYMMDD
21 YYYY-MM-DD
22 YYYY/MM/DD
23 YYYY.MM.DD
24 YYYY MM DD
25 YYYYMMDD
26 MM-DD
27 MM/DD
28 MM.DD
29 MM DD
30 MMDD
31 DD-MM
32 DD/MM
33 DD.MM
34 DD MM
35 DDMM
By default, the date format depends upon the Posting Protocol selected.
Use following command to program date fill flag for date field:
8170-Date Fill Flag
Where,
0 Disable
1 Enable
Time
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
1 HH:MM:SS
2 HH:MM
Use following command to program time fill flag for time field:
8171-Time Fill Flag
Where,
0 Disable
1 Enable
Duration
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
1 HH:MM:SS
2 HHMMSS
3 Minutes
4 Seconds
Use following command to program duration fill flag for duration field:
8172-Duration Fill Flag
Where,
0 Disable
1 Enable
Units
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is 2.
1 Higher Currency
2 Lower Currency
Use following command to program the amount fill flag for amount field:
8173-Amount Fill Flag
Where,
0 Disable
1 Enable
Currency Symbol
Use following command to program column position for currency symbol field:
8141-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 000.
Use following command to program field length for currency symbol field:
8142-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 001.
Alignment Meaning
1 Left Alignment
2 Right Alignment
Use following command to program fill character for currency symbol field:
8144-Fill Character
Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Space'.
If currency string/symbol to be used is fewer than 8 characters, terminate the command with #*.
Use following command to program column position for call type indicator field:
8146-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 000.
Use following command to program field length for call type indicator field:
8147-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 001.
Use following command to program alignment for call type indicator field:
8148-Alignment
Alignment Meaning
1 Left Alignment
2 Right Alignment
Use following command to program number string for call type indicator field:
8149-Number Index-1-Number String
Where,
Number Index is from 01 to 36.
Number String is of four digits. Terminate with #* if the string is less than four digits.
Use following command to program text string for call type indicator field:
8149-Number Index-2-Text String
Where,
Number Index is from 01 to 36.
Text String is a string of Alphanumeric characters. Terminate with #* if the string is less than four digits.
Keep the Text String same as the Field Length.
By default, all the entries in this table are blank.
Called Location
Use following command to program column position for called location field:
8150-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 000.
Use following command to program field length for called location field:
8151-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 005.
Alignment Meaning
1 Left Alignment
2 Right Alignment
Called Number
Use following command to program column position for called number field:
8154-Column Position
Where,
Column Position is from 000 to 128.
Use following command to program field length for called number field:
8155-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 019.
Alignment Meaning
1 Left Alignment
2 Right Alignment
Use following command to program number format for called number field:
8157-Number Format
Where,
1 Continuous
2 Separated
Account Code
Use following command to program column position for account code field:
8158-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 000.
Use following command to program field length for account code field:
8159-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 004.
Alignment Meaning
1 Left Alignment
2 Right Alignment
Use following command to program fill character for account code field:
8161-Fill Character
Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Space'.
Code Meaning
0 No
1 Yes
Remarks
Alignment Meaning
1 Left Alignment
2 Right Alignment
001 041
002 051
: :
200 096
001 India
002 Kenya
: :
200 UAE
Relevant Topics:
1. “Communication Ports” 1038
2. “Call Cost Calculation (CCC)” 887
3. “Station Message Detail Recording” 1472
4. “Station Message Detail Recording-Storage” 1545
5. “Station Message Detail Recording-Online” 1473
6. “Station Message Detail Recording-Report” 1531
7. “Call Duration Control (CDC)” 919
What's this?
• The ETERNITY can generate report for the calls which were stored in the buffer when it is required by the
user. This is called SMDR Reports feature.
• To get the online report SE should assign the destination port for IC calls, Internal calls and OG calls. SE
should ensure that the storage flag is enabled to store the calls in the buffer of the system. Refer chapter
“Station Message Detail Recording-Storage” to enable the storage of calls.
• The ETERNITY supports to change the default filters for getting the SMDR Report, such as report for the
calls with specific call duration or for the calls only for specific originating stations or for the calls with
specific dates on which the calls were made or received etc. The SA can configure these filters as per
user's requirement.
• The ETERNITY provides flexibility to user, to get the report using following options:
• Refer the sample reports at the end of chapter for IC, OG and Internal calls with information about called
number, calling number, date of call, time of call, port or trunk number on which call was established and
speech duration, etc.
How to configure
Code Meaning
0 None
1 COM1
2 COM2
3 Printer Port
4 Ethernet Port
If you assigned Ethernet Port as destination port, use the following command to assign the IP Address to the
Ethernet Port:
2934-IP Address
By default, IP Address is 192.168.1.104
• On assigning the communication port or the printer port or Ethernet Port as the destination port, the
report generation is directed to the port assigned.
• Care should be taken by the SE not to overlap the assignment of same ports to the different processes.
• By default, the port assigned is None. This means the Report printing is disabled.
Filter Commands:
The SA can program the filters as per user's need to generate a customized report. These commands enable the
user to select the type of call reports generated viz. Normal calls, DID calls, Unanswered calls, Long Speech
Duration calls. It is also possible to generate call reports for a range of stations (SLT, DKP), different type of trunks
(TWT, BRI, T1E1PRI etc.).
Filter Commands
To set filter to print all calls with specific call duration 1072-157-Seconds
To set filter to print all calls with specific call duration 1072-157-Seconds
To set filter to print all call with specific Unanswered duration 1072-158-Seconds
To set filter to print all calls with specific hold duration 1072-159-Seconds
To set filter to print all IC calls received by the station 1072-160-Flexible No.-Flexible No.
To set filter to print all IC calls from numbers matching in a Number 1072-169-Number List
list
Set OG print filter to print the calls terminated from the LD trunk. 1072-116-LD-LD
Set print filter to print all IC calls received from LD trunk. 1072-117-LD-LD
Set OG print filter to print the calls originated from the LD trunk. 1072-118-LD-LD
Where,
Flag Meaning
0 Don't store
1 Store
Seconds = 000-255.
TWT = 000-128.
BRI = 00-32.
T1E1PRI = 0-8.
E&M = 000-128.
Mobile = 00-64.
SIP = 00-32.
Date = 01-31
Month = 01-12.
Year = 0000-9999.
Number List = 00-16.
For example, for TWT filter, range can be 000 to 128, as explained below:
• 000-000: The report of calls will be printed which are made from any TWT trunk port.
• XXX-XXX: Calls will be printed from XXX-XXX. TWT number, where XXX = 001 to 128.
• 000-128: Error tone (Not valid).
Same explanation is true for other types of port like BRI, T1E1PRI, E&M and Mobile.
For Number List, range can be 00-16. Where, if 00 is set, no Number list is checked while processing the call.
Flag Meaning
0 Abort
1 Start
Once the Start command is issued, the report generation stops only after the complete report based on the filters
set is generated. ETERNITY provides a facility to abort the report generation in midway (1072-171-0). Once the
report generation is aborted, then it has to be explicitly started (1072-171-1) if the report is required again.
Flag Meaning
Daily Reports:
Use following command to program the time for daily scheduled reports:
1072-173-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00.
Weekly Reports:
Use following command to program the day and time for weekly scheduled reports:
1072-174-Day-HH-MM
Where,
HH and MM is in 24 hour format.
Day is from 1 to 7 (1 is Sunday, 2 is Monday ……… and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports:
ETERNITY provides a facility to Abort the scheduled report generation midway (119-171-0). This aborts
the current report generation but does not affect any other scheduled report generation. The abortion of a
report does not affect its consecutive schedule. That is if a daily report is aborted on Monday, it does affect
the report generation schedule of Tuesday.
Use following command to assign destination port for Online SMDR-Internal Report:
2831-Code
Where,
Code Meaning
0 None
1 COM1
2 COM2
3 Printer Port
4 Ethernet Port
If you assigned Ethernet Port as destination port, use the following command to assign the IP Address to the
Ethernet Port:
2834-IP Address
By default, IP Address is 192.168.1.104
On assigning the communication port or the printer port or Ethernet Port as the destination port, the report
generation is directed to the port assigned.
SE should take care, not to overlap the assignment of same ports to the different processes.
By default, the port assigned is None. This means the Report printing is disabled.
Filter Commands:
Filter Command
To set filter to print all internal calls from a station. 1072-137-Flexible No.-Flexible No.-
Type
To set filter to print all internal calls with duration greater than 1072-138-Seconds
specified.
Where,
Seconds = 000-999.
Type Meaning
Flag Meaning
0 Abort
1 Start
By default, flag is 0.
Once the start command is issued, the report generation stops only after the complete report based on the filters
set is generated. ETERNITY provides a facility to abort the report generation in midway (1072-141-0). Once the
report generation is aborted, then it has to be explicitly started (1072-141-1) if the report is required again.
SA should issue the command to start the report, automatically at a particular time (Daily) or at a particular time on
a particular day of the week (Weekly) or at a particular time on a particular date of a month (Monthly).
Once these parameters are programmed and the report generation is enabled, there is no need of any initiation
command.
Flag Meaning
Daily Reports:
Use following command to program the time for daily scheduled reports:
1072-143-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00
Weekly Reports:
Use following command to program the day and time for weekly scheduled reports:
1072-144-Day-HH-MM
Where,
HH and MM is in 24 hour format.
Day is from 1 to 7 (1 is Sunday, 2 is Monday ……… and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports:
Use following command to program the date and time for monthly scheduled reports:
1072-145-Date-HH-MM
Where,
HH and MM is in 24 hour format.
Date is from 01 to 31.
By Default, HH:MM is 10:00 and Date is 01.
ETERNITY provides a facility to abort the scheduled report generation in midway (1072-141-0). This
aborts the current report generation but does not affect any other scheduled report generation. The
abortion of a report does not affect its consecutive schedule. That is if a daily report is aborted on Monday,
it does affect the report generation schedule of Tuesday.
OG Calls Report
Destination Port
Code Meaning
0 None
1 COM1
2 COM2
3 Printer Port
4 Ethernet Port
• On assigning the communication port or the printer port or Ethernet Port as the destination port, the
report generation is directed to the port assigned.
• Care should be taken by the SE not to overlap the assignment of same ports to the different processes.
• By default, the port assigned is None. This means the Report printing is disabled.
Filter Commands:
SA can configure various filters to generate a report as per the user's requirement. It is possible to program the
following filters:
Filter Commands
To set filter to print all calls terminated on T1E1PRI 1072-105- T1E1PRI - T1E1PRI
To set filter to print outgoing calls for department billing group 1072-109-Group No.-Group No.
To set filter to print all calls of numbers matching in a Number 1072-112-External Number List
list
To set filter to print all calls duration more than specified 1072-113-Seconds
To set filter to print all calls with units more than specified 1072-114-Units
To set filter to print all calls made using Account Code 1072-115-Account Code-Account Code
To set filter to print all calls originated on stations 1072-102-Flexible No.-Flexible No.
To set filter to print calls originated on TWT 1072 - 183 - TWT - TWT
To set filter to print calls originated on BRI 1072 - 184- BRI- BRI
To set filter to print calls originated on T1E1PRI 1072 - 185 - T1E1PRI - T1E1PRI
To set filter to print calls originated on E&M 1072 - 186 - E&M- E&M
To set filter to print calls originated on Mobile 1072 - 187 - Mobile- Mobile
To set filter to print calls originated on SIP 1072 - 188 - SIP- SIP
Where,
Seconds = 000-999.
Number List = 01-16.
TWT = 001-128.
BRI = 01-32.
T1E1PRI = 1-8.
E&M = 001-128.
Mobile = 00-64.
SIP = 00-32.
Date = 01-31.
Month = 01-12.
Year = 0000-9999.
Unit = 0000-9999.
Account Code = 0000-9999.
For Account Code filter, range can be set as various options as explained:
• 0000-0000: In report, all calls will be printed which doesn't consist of account code and calls with account
codes will not be printed.
• XXXX-XXXX: In report, calls with account code XXXX to XXXX will be printed, where XXXX = 0001 to
9999.
• 0000-XXXX: In report, all calls will be printed which are without account code and which have account
code upto XXXX
where,
XXXX=0001 to 9999.
Thus, if XXXX = 0020, then in report all calls without account code and with account code 0001 to 0020 will
be printed.
• 000-000: Valid range. I report calls will be printed which are made from any TWT trunk port.
• XXX-XXX: Calls will be printed from XXX-XXX. TWT number, where XXX = 001 to 128.
• 000-128: Error tone (Not valid). Same explanation is true for other types of port like BRI, T1E1PRI, E&M
and Mobile.
For Number List, range can be 00-16. Where, if 00 is set, no Number List is checked.
How to set filter to print the calls department bill group wise from SA mode?
Use following command to set filter to print outgoing calls for department billing group
1072-109-Department Bill Group-Department Bill Group
Where,
Department Bill Group is from 00 to 99.
If command is used for more than one group, then the calls of first department billing group will be printed followed
by next department group.
By default, first Department Bill Group is '00' and second Department Bill Group is '99'.
Calls made by all stations, all TWT, ALL BRI, ALL E&M and All T1E1PRI are printed.
The Date range = 01-05-2005 to 31-12-2100.
The Time range = 00:00-23:59.
The External Number list = 02.
Units = 0000.
Seconds = 000.
Please note that the date format depends on the date format of the system.
The ETERNITY supports Deletion of SMDR Calls (From SA Mode) made from a particular station or a range of
stations.
Use the following Command to delete the calls made by a station or a range of stations:
1072-131-Flexible Number-Flexible Number
Use the following Command to delete the calls made on a Date or from a Date:
1072-132-DD-MM-YYYY-DD-MM-YYYY (The format of the date depends on the date format of the system)
Flag Meaning
0 Abort
1 Start
By default, Flag is 0.
Once the start command is issued, the report generation stops only after the complete report based on the filters
set is generated. ETERNITY provides a facility to abort the report generation in midway (1072-121-0). Once the
report generation is aborted, then it has to be explicitly started (1072-121-1) if the report is required again.
Flag Meaning
Daily Reports:
Use following command to program the time for daily scheduled reports:
1072-123-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00.
Weekly Reports:
Use following command to program the day and time for weekly scheduled reports:
1072-124-Day-HH-MM
Where,
HH and MM is in 24 hour format.
Day is from 1 to 7 (1 is Sunday, 2 is Monday ……… and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports:
Use following command to program the date and time for monthly scheduled reports:
1072-125-Date-HH-MM
Where,
HH and MM is in 24 hour format.
Date is from 01 to 31.
By Default, HH:MM is 10:00 and Date is 01.
ETERNITY provides a facility to abort the scheduled report generation in midway (1072-121-0). This
aborts the current report generation but does not affect any other scheduled report generation. The
abortion of a report does not affect its consecutive schedule. That is if a daily report is aborted on Monday,
it does affect the report generation schedule of Tuesday.
Relevant Topics:
1. “Station Message Detail Recording” 1472
2. “Station Message Detail Recording-Storage” 1545
3. “Station Message Detail Recording-Online” 1473
4. “Station Message Detail Recording-Posting” 1489
The Offline report for Internal calls looks like shown below:
The Off line report for Outgoing calls looks like shown below:
What's this?
• ETERNITY supports Station message detail recording (SMDR) facility for Incoming calls, Outgoing Calls
and Internal Calls. Refer chapter “Station Message Detail Recording”.
• The storing of each type of call is allowed only if it is enabled by the relevant command.
• ETERNITY will store the Incoming, Outgoing and Internal calls in SMDR, as per programming of SMDR
storage filters. These filters can be set by using specific commands mentioned in this chapter.
• The call records are stored in the SMDR buffer. Once the SMDR buffer is full, the next call is stored in
place of the oldest call in the SMDR buffer. This is known as First In First Out (FIFO) logic.
• The SMDR buffer data is maintained even during power failures. However it is advisable to take frequent
printouts of the calls to avoid accidental loss of the data.
• SMDR-IC Storage
• SMDR-Internal Storage
• SMDR-OG Storage
SMDR-IC Storage
• The SMDR facility is used to keep track of the incoming calls received by the ETERNITY. The SMDR is
useful in monitoring and managing the incoming traffic in more effective manner. The ETERNITY can keep
log of each call received by it in the SMDR buffer. Each incoming call is stored in the form of a record in the
buffer.
• The SA or SE can clear the SMDR IC buffer using appropriate command and password from the SA mode.
SMDR-Internal Storage
• SMDR-Internal storage is used to keep record of calls made from one station to other within the
organization. The ETERNITY can keep track of each call made by any station. Each call is stored in form
of a record in the SMDR buffer.
• Maximum 1000 records of Internal calls can be stored in the SMDR buffer.
• The SA can print these calls on a printer or can even transfer this data on a computer for storage or further
analysis.
• If BRI's access code is sent as "Calling Number" it will be stored in the "CALLING STATION" in SMDR
Internal Calls Report.
• If BRI's access code is dialed or ISDN Terminal's flexible number is dialed by any station or another ISDN
Terminal it will be stored in the "CALLED STATION" field of the SMDR Internal Calls Reports.
SMDR-OG Storage
• SMDR-OG storage facility is used to keep records of calls made from the system. This SMDR is useful in
monitoring and controlling the cost of telephone calls. The ETERNITY can keep track of each call made by
any station.
• The SE or SA can clear the SMDR-OG buffer by using appropriate password from the SA mode. The
System Administrator (SA) can print these calls on a printer. The SA can even transfer this data on a
computer for storage or further analysis.
• In case of a call transfer from one station to another station, the system stores two calls with same number
but different duration for both the stations. This is called 'Call Toggle'. Refer chapter “Call Toggle” for more
details.
• If ISDN Terminal doesn't send its flexible number while making OG call, the BRI port's access code will be
stored in the column "STN" in SMDR OG Calls Report.
How to configure
SMDR-IC Storage
Enable SMDR-IC Storage:
This command enables/disables the storage of incoming calls depending upon the storage filters set.
By default, all the calls are stored as per the filters set.
Filter commands:
These commands enable the user to select the type of calls to be stored viz. All calls, Trunk wise calls, Un-
answered calls, DID calls, etc. Calls will be stored only if duration, exceeds than the set value.
Where,
Seconds = 000-999.
Flag Meaning
0 Don't store
1 Store
SMDR-Internal Storage
Enable SMDR-Internal Storage:
This command enables/disables the storage of internal calls depending upon the storage filters set.
By default, all the calls are stored as per the filters set.
Filter commands:
Filter Command
Where,
Seconds = 000-999.
By default, the speech duration is set to 000.
SMDR-OG Storage
Enable SMDR-Internal Storage:
This command enables/disables the storage of outgoing calls depending upon the filters set.
Use following command to set SMDR storage flag:
2701-Storage Flag
Where,
Filter commands:
These commands enable the user to select the type of calls to be stored viz. destination number wise, duration
wise or cost wise.
It is possible to store outgoing calls selectively depending on the destination numbers. The ETERNITY supports
this feature in association with Number Lists. An outgoing call will be stored only if the number matches with an
entry in the Number List assigned.
Use following command to assign a Number list containing numbers for call storage:
2702-Number List
Where,
Number List is 01-16.
By default, Number List assigned is 02.
Duration wise:
Sometimes it is required to filter out the calls of small durations. System will not store the calls with duration less
than duration, programmed.
Unit wise:
Sometimes it is required to filter out the calls based on Call units. System will not store the calls with units less than
the units programmed:
Call Toggle:
0 Toggle OFF
1 Toggle ON
0 OFF
1 ON
When Call Toggle Flag is set OFF, then only above command (2717) can be effective. Refer chapter “Call
Toggle” for more details.
Relevant Topics:
1. “Station Message Detail Recording” 1472
2. “Station Message Detail Recording-Report” 1531
3. “Station Message Detail Recording-Online” 1473
4. “Station Message Detail Recording-Posting” 1489
What’s this?
The ETERNITY monitors all its activities and maintains records of these activities in the System Activity Log.
The System Activity Log has a buffer capacity of 250 records. The Activity Log stores records using the FIFO
method.
This log can be printed on a printer or downloaded on a computer in form of a report. The activity log can be printed
or downloaded in two modes:
• Report (Offline): The activity report is printed/downloaded whenever desired. In the Offline mode, the last
250 activities recorded by the system are printed/downloaded.
The System Administrator can print/download System Activity Log, online or offline using any serial device
connected to the COM Port of ETERNITY.
ETERNITY also supports Syslog Client for System Activity Logs. The Syslog Client enables the system to send
activity logs in syslog format to the remote ‘Syslog Server’. You can view the logs on the remote server.
You may use Syslog for System Activity Log, if you have no spare COM Port on your ETERNITY or your system
does not have a COM Port, as in ETERNITY PE3SS.
How it works
• A destination port, serial or ethernet, must be assigned for activity logs. The system will send the activity
log to this port.
• If the System Administrator extension is a DKP or an Extended IP Phone, a DSS Key can be assigned for
System Activity Log.
• Whenever an activity is recorded by the system, the DSS key, if assigned for this feature on the System
Administrator’s DKP/Extended IP Phone extension, is turned ON.
• The System Administrator can view the activity log by pressing the DSS key (if assigned). The DKP/
Extended IP Phone of the System Administrator will display the activity in this format:
The format of the Date will be DD-MM or MM-DD as per Date Format selected in the Real Time Clock
settings of the system.
Index Activity
3 DD-MM-YYYY HH:MM:SS Card Present: Slot=SS, Type: <ZZZZZ > Card V02R01
20 DD-MM-YYYY HH:MM:SS BITE Test of Slot No. xx Port Offset xxx : Pass
21 DD-MM-YYYY HH:MM:SS BITE Test of Slot No. xx Port Offset xxx : Fail
23 DD-MM-YYYY HH:MM:SS VMS Disk is full (i.e. Pen Drive of VMS Card is full).
SS-Slot Number, PP -Port Number, TT -Trunk Number, XXXX -Station Number, v-Version for VvRr, r-
Revision for VvRr, ZZZZ = Card Type
When installed in the Hotel mode, the ETERNITY captures Hotel-Motel Activity Log. To know more, see the
ETERNITY Hospitality System Manual.
How to configure
The two functional parts of system activity log are: Storage and Report Generation in the Online or Report modes.
To be able to use this feature, you must enable storage of Activity Logs, and assign the Syslog Server address as
Destination Port for the logs.
If you want to use Syslog Server, you must assign the IP Address of the remote Syslog server as the Destination
Port for the System Activity Log.
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System
Activity Log.
For instructions on configuring a DSS key on a DKP, see “DSS Keys Programming”.
For instructions on configuring a DSS key on an Extended IP Phone, see “Matrix Extended IP Phone Settings”
under Configuring SIP Extensions.
• To generate System Activity Log - Online, i.e. as and when the activity occurs, select Destination Port.
for Online SAL from the following options:
• Comm. 1 or Comm. 2: Select a communication port if you want to use a serial device to capture the
logs. Make sure the device is connected to the COM port.
• Printeries printer port is available on ETERNITY ME only. Select this option if you want the logs to be
sent to the printer port of ETERNITY ME.
• Ethernet: Select Ethernet port if you want to use the remote syslog server for the logs.
• If you select Ethernet, in the Destination IP Address: Port - Online SAL field, enter the IP
Address and the port of the remote Syslog Server.
• To generate System Activity Log - Report, i.e. offline, whenever desired, select Destination Port for SAL
Report from the options Comm. 1 or Comm. 2, Printer, Ethernet. Default: None.
• If you selected Ethernet as the Destination port, in the Destination IP Address: Port - SAL Report
field, enter the IP Address and the port of the remote Syslog Server.
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
• 6404-IP Address
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
• 6406-IP Address
How to use
You can start and stop System Activity Log - Online and Report from the System Administrator mode using Jeeves
or dialing SA Commands from an extension phone.
• Open Jeeves.
• To clear System Activity Logs from the buffer, click the Clear System Activity Log button.
• Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
The Online System Activity Log report would look like this:
The Offline System Activity Log report would look like this:
What’s this?
The ETERNITY provides a facility to display the last activity monitored by the system on the System Administrator’s
extension phone.
How to use
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Activity Log Display.
For instructions on configuring DSS Keys on a Digital Key phone, see “DSS Keys Programming”.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see “Matrix Extended IP Phone Settings”
under Configuring SIP Extensions.
• Go Off-hook.
• Press the DSS key assigned to System Activity Log Display.
OR
• Dial 1072-009
• The last recorded Activity log appears on your phone’s display in this following format: Date-Time-Activity
Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 23.
See System Activity Log Activity Index table in “System Activity Log”.
The date and month format will be DD-MM or MM-DD as per date format set in the system. See “Real Time
Clock (RTC)” for setting the date format.
What’s this?
The ETERNITY provides a facility to display the last fault monitored on the system on the System Administrator’s
extension phone.
How it works
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Fault Log.
• When a fault occurs, the LED of the DSS Key assigned for the System Fault Log, glows.
• The System Administrator may press the DSS key or dial the System Fault Log feature access code to
acknowledge it.
• On pressing the DSS Key or dialing of the acknowledgment command, the LED of the Fault Log key is
turned OFF.
How to use
• Go Off-hook.
• The Fault log appears on your phone’s display in this format:Date-Time-Fault Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 12.
See System Fault Log Activity Index table in “System Fault Log”.
What’s this?
The ETERNITY maintains a log of all system faults. The system Fault Log has a buffer capacity of 100 records. The
Fault Log stores records using the FIFO method.
The System Fault log can be printed on a printer or downloaded on a computer in form of a report. The report can
be printed or downloaded by the System Administrator in two modes:
• Report (Offline): The faulty report is printed/downloaded whenever desired. In the Report (Offline) mode,
the last 100 faults recorded by the system are printed/downloaded.
The System Administrator can print/download System Fault Log, Online or Report using any serial device
connected to the COM Port of ETERNITY.
Matrix ETERNITY also supports Syslog Client for System Fault Logs. The Syslog Client enables the system to
send fault logs in syslog format to the remote ‘Syslog Server’. You can view the logs on the remote server.
You may use Syslog for System Fault Log, if you have no spare COM Port on your ETERNITY or your system does
not have a COM Port, as in ETERNITY PE3SS.
How it works
• A destination port for sending the report must be selected to which the system can send the log.
• If the System Administrator extension is a DKP or an Extended IP Phone, a DSS Key can be assigned for
System Fault Log.
• Whenever a fault is detected, the LED of the Fault Log DSS key, if assigned, is turned ON. The buzzer of
the Master Card of ETERNITY ME starts sounding.
• If more than one DKP/Extended IP extension is assigned Fault Log DSS Key, the LED of all keys will be
turned ON.
• The System Administrator must acknowledge the Fault indication by pressing the Fault Log key or by
dialing the Fault Log access code. The LED of the Fault Log key is turned OFF. The buzzer of the Master
Card of ETERNITY ME stops sounding.
The different activities that are logged into system fault log are summarized in this table:
2 DKP Absent, Flexible No. of the port, Slot No. <>, Port No<>
3 SLT Short, Flexible No. of the port, Slot No. <>, Port No <>
4 SLT Open, Flexible No. of the port, Slot No. <>, Port No. <>
9 RTC Failure
10 VoIP LAN Lost, Slot No. <>, Port No. <> on DD-MM-YYYY at HH:MM:SS
VOPP Fail: If VoPP fail message is logged, the ETERNITY VoIP card will not be functional.
Registration Timer Fail: The system may fail to load either the Re-registration Timer or the Registration
Retry Timer. In such a case the Proxy SIP trunk will remain un-registered and will not be functional.
The system will decode the registration status message received from ETERNITY VoIP module and, if it is
found to be a problem caused by Registration Timer Failure, this will be logged in the System Fault Log.
This can happen to one or more SIP trunks, while the other SIP Trunks are functioning normally. You need
to restart the system to resolve the problem.
How to configure
To be able to use this feature, you must enable storage of Fault Logs, and assign a Destination Port for the Fault
Logs. The destination port may be a COM Port, Ethernet Port, or Printer port (available on ETERNTIY ME only).
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System Fault
Log.
For instructions on configuring DSS Keys on a Digital Key phone, see “DSS Keys Programming”.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see “Matrix Extended IP Phone Settings”
under Configuring SIP Extensions.
You may configure the System Fault Log settings using Jeeves and by dialing system commands from a telephone
connected to the ETERNITY.
• Scroll to the System Fault Log link and click this link.
• To generate System Fault Log - online, i.e. as and when the fault occurs, select the Destination Port for
Online SFL from the following options:
• Comm. 1 or Comm. 2: Select a communication port if you want to use a serial device to capture the
logs. Make sure the device is connected to the COM port.
• Printer: The printer port is available only on ETERNITY ME. Select this option if you want the logs to
be sent to the printer port of ETERNITY ME.
• Ethernet: Select Ethernet port if you want to use the remote syslog server for the logs.
• If you select Ethernet, in the Destination IP Address: Port - Online SFL field, enter the IP Address
and the port of the remote Syslog Server.
• To generate System fault Log - Report, i.e. offline, whenever desired, select Destination Port for SFL
Report from the options: Comm. 1 or Comm. 2, Printer, Ethernet. Default: None.
• If you selected Ethernet as Destination Port, in the Destination IP Address: Port - SFL Report field,
enter the IP Address and the port of the remote Syslog Server.
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
• 6455-IP Address
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
• 6457-IP Address
• Exit SE Mode.
• Open Jeeves.
• To clear System Fault Logs from the buffer, click the Clear System Fault Log button.
• Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
Blank 00
Time (HH:MM:SS) 12
Event Description 24
What’s this?
System Parameters are general parameters, related to features and facilities that are applied system-wide, such as
customer name, Day-Night mode, storage of call logs, end of dialing, alarms, DID call disconnect options,
Presence, and DND messages. Each of these is described briefly here.
Customer Name
You can assign the name of the enterprise/organization that is using ETERNITY as the Customer Name. The
Customer Name may contain up to 80 characters. You may enter the address of organization/enterprise along with
the name.
The Customer Name you assign will appear on the various System Reports generated and printed by the
ETERNITY.
System Parameters
• Customer Profile: ETERNITY two major applications: Enterprise application to meet requirements of
businesses, and Hospitality application to meet the specific requirements of Hotels and Hospitals.
You must select Customer Profile as Enterprise or Hotel according to the application you are using. When
you select the Customer Profile, all the features and facilities specific to the application Enterprise/Hotel
along with their default settings are loaded. By default, the Customer Profile of ETERNITY is defined as
'Enterprise'.
• Onsite configuration: This flag is enabled, the configuration GUI of ETERNITY, Jeeves, will show the
pages for only those trunks and extension port types that are on board the system, i.e., detected by the
system at Power-On.
• Voice Mail Group Destination: This is the group of extensions to be assigned as Voice Mail group.
If you have interfaced an external Voice mail System or Auto Attendant with the ETERNITY, you must first
create a routing group of the SLT extension ports connected to the VMS. Assign the same routing group
number in this field.
If you have installed the ETERNITY VMS Card, the system will automatically assign routing group number
‘96’ reserved for the VMS Group. Default: 96.
• External Voice Mail RTC Synchronization: When you interface an external Voice Mail System with the
ETERNITY, you need to synchronize the Real Time Clock (RTC) of the external VMS device and the RTC
of ETERNITY to ensure precision in the operation of time-based VMS features like Message Wait,
Greeting Messages, Call Transfer.
Enable the External Voice Mail RTC Synchronization, if you have connected an external Voice Mail
System to the ETERNITY.
• Station Name Pattern: The Station Name Pattern is the format in which the names of extensions will be
stored on the extension phones and displayed to other extensions. You can store names by First Names
only, First names and Last Names. You can also add Titles indicating gender, designation, rank, social
standing, like Mr. Mrs. Ms., Dr., Prof. Cmdr., Rev., to Names of extensions.
Option Meaning
3 Name only
4 First Name<space>Name
5 Title<space>Name
6 Title<space>Name
Station Name Pattern must be configured for the Guest Name and Title feature of the ETERNITY
Hospitality module. To know more, refer the feature description in the ETERNITY Hospitality System
Manual.
By default, Name Only is selected as the Station Name Pattern when ETERNITY is operated in the
Enterprise mode, and Title<space>Name is selected as the Station Name Pattern when ETERNITY is
operated in the Hotel Mode.
• Global Hold: This parameter is related to the “Call Hold” feature of ETERNITY. Global Hold enables you
pick up a call that has been put on hold by any extension from a DKP/Extended IP Phone extension of
ETERNITY.
• Store Internal Calls in Missed Call Log: ETERNITY stores call logs of external calls only. You may
instruct the system to store internal calls in the Missed Call log by enabling this parameter. See “Call
Logs”.
• Store Internal Calls in Dialed Call Log: You may instruct the system to store internal calls in the Dialed
Call log by enabling this parameter. See “Call Logs”.
• Store Internal Calls in Answered Call Log: You may instruct the system to store internal calls in the
Answered Call log by enabling this parameter. See “Call Logs”.
• MoH Source when Trunk kept on Hold: You can select the source of music on hold (MoH) that
ETERNITY should play to external callers who are put on hold. Select Internal (VM-01) if you want the
system to play MOH recorded in the Voice Module. Select External (AIP), if you want the system to play
MOH from an external device connected to the Analog Input Port (AIP) of the system.
• End of Dialing Digit: End of Dialing Digit is a single digit, on receipt of which, the system considers the
number string dialed by the extension users as the complete string. It does not wait for further digits to be
dialed, and dials out the number. The digits * (star) or # (hash/pound) are used to indicate end of dialing to
the system, as these are unique and distinguishable from the digits generally dialed by extensions (0, 1,
2... to 9). Default: #
• Give Off-hook Alert to Operator: When this flag is enabled, the system will detect extensions that are off-
hook and ring on the Operator extension to alert the Operator about the state of the phone. This alert is
useful for detecting whether the handset of extension phones are placed correctly. Read the feature
description for “OFF-Hook Alert” to know more.
• Day/Night Mode: Certain features of the ETERNITY require extensions and trunks to behave differently
according to the working hours, break hours and non-working hours, which are referred to as Time Zones.
The Time Zones, i.e. working hours, break hours and non-working hours, are defined for the entire week in
a Time Table. Time Table is assigned to trunks, extensions and other time-zone dependant features.
You can set the Time Zone of the system as Working-Hours or Non-Working hours any time you want by
setting the system in the Day Mode or the Night Mode, or let the system operate as per the Time Table
assigned. For more details see “Day Night Mode” and “Time Tables”.
• Emergency Dialing Reporting: When this flag is enabled, the system detects the extension that has
made the emergency call and reports it to the Operator extension. Thus the Operator can know which
extension has made an emergency call. See “Emergency Detection and Reporting” to know more.
• Replace '+' from CLI: The Mobile network presents the calling party number with prefix '+' to the called
party. ETERNITY allows you to remove '+' and replace it with an appropriate number string as required. To
do this, you must enable Replace '+' from CLI.
You may also program the number string with which ‘+’ is to be replaced in the CLI in the Replace '+' from
CLI with the number string field. In this field, enter the number string with which you want to replace '+'
received as prefix of calling party number.
If you keep the number string field blank, ETERNITY will remove '+' sign from the CLI of calling party and
present the remaining digits on the CLI of the Called Party.
For example:
The number string +919327237228 is received as CLI.
If ‘00’ is configured as the replace string, the CLI number would become 00919327237228
If no replacement string is configured (i.e. left blank), the CLI number would be presented as
919327237228.
• Disconnect DID call, when Dialed Number is not Responding: when this flag is enabled the system
disconnects the DID call if there is no reply from the landing destination extensions. The DID call will not be
routed to the Operator. Default: disabled.
• Disconnect DID call, when Caller Does not Dial any Digit: when this flag is enabled, the system will
disconnect the DID call if the caller fails to dial a digit within the First Digit Wait Timer. The DID call will not
be routed to the Operator. Default: disabled.
• Play Beep when Conference/Dial-in Conference Starts: This is a common flag for the features
“Conference-Multiparty”, “Conference Dial-In”, “Emergency Conference”, and “Raid”. When this flag is
enabled,
• the system plays a warning beep to the extension which is being raided by another extension, before
establishing three-way speech.
• the system plays beeps to the other participants in a Dial-In Conference when a new participant joins in
(i.e. dials in to an on-going Dial-In Conference)
• the system plays beeps to the other participants connected in a Multi-Party Conference and an
Emergency Conference, when a new participant is included.
If you disable this flag, no warning Beep will be played in Raid, the existing participants in a Dial-In or Multi-
party conference will not hear any beep tone indicating the addition of a new participant.
• Play Beep when Call Taping and Conversation Recording Starts: This is a common flag for the
features “Call Taping” and “Conversation Recording”. When this flag is enabled, the system plays Beeps to
the extensions/calling party and extension before it starts taping the call in the common mailbox or
recording the conversation in the extension’s mailbox.
When this flag is disabled, no indication will be given to the opposite party when the call is being taped/
conversation is being recorded. Default: Enabled.
• Play Feature Tone in place of Dial Tone when Call Forward is Set: You can select whether the you
want the system to play Feature Tone instead of Dial Tone to the extensions when Call Forward is set on
these extensions. When this flag is disabled, the system will play dial tone to the extension on which Call
Forward is set, whenever the extension goes Off-hook. Default: Enabled.
• Call Proceeding Tone - Multistage Dialing: This flag is used in “Multi-Stage Dialing” where you need to
configure Pause and Wait for Answer in the Substitute Number string for the number string dialed by the
extension users.
When an extension user makes a call using a Calling Card, and the system dials out the number in stages,
the extension user will get Ring Back Tone twice; first after the system has dialed the Calling Card
Number, and again after the system has dialed out the destination number (called party number). Thus the
extension user will get Ring Back Tone, twice. To avoid this, you may configure the 'Call Proceeding Tone'
to be played by the system when using Multi-Stage Dialing.
• Network Tone: If this option is selected, the extension user will get Ring Back Tone after dialing the
calling card number and again, after the system has dialed the called party number (when the system is
dialing out the number with Pause and Wait for Answer configured in the substitute number string).
• Pseudo Tone: If this option is selected, the extension user will get Feature Tone when the user has
completed dialing all the digits. At the end of the tone, the extension user gets connected to the called
party (destination number).
• Silent: If this option is selected, the extension user will get Silence (no tone), after the extension user
has completed dialing all digits. After dialing out the called party number in DTMF, the system will
connect the caller to the called party number (destination number).
• Companding Algorithm: The companding Algorithm —A law or law—is automatically selected when
you select Region for ETERNITY. However, if necessary, you may change the default companding
Algorithm that appears in this field. Select the companding Algorithm according to the Regulatory
Requirement of the country where ETERNITY is installed.
• Language of SE, SA and Front Desk User Web Interface: The GUI of ETERNITY supports the
languages English, Italian, Spanish, French, German, and Portuguese. When you select ‘Region’ for
ETERNITY, one of these languages will be applied as appropriate for the region you selected. For
instance, if you selected India, English will be applied. If you selected Spain, Spanish will be applied. If you
selected a country where none of these languages are the local language, English will be applied.
The language set by the system on Region selection will be applied on the pages of the GUI for every login
session. You can change the default language set on Region selection, by configuring this parameter.
• Form Feed in Report Printing: To print each system report on a separate page, enable Form Feed in
Report Printing. Default: Enabled.
• Minimum No. of digits received in CLI to consider the call is from Public N/w: This parameter is used
for distinguishing between incoming calls from Public Network and Private Network on the basis of the
number of digits received in the CLI.
This parameter is applicable to calls originating on the T1E1PRI ports (Tie-Line) configured for the Q-
Signaling. By default, CLI number with 8 or more digits will be considered as call from Public Network.
ETERNITY will check this parameter, whenever the incoming call is to be analyzed as call from PISN
(Private Integrated Subscriber Network) or non-PISN (Public Network Number).
• Mailbox for Call Taping: When “Call Taping” feature is applied on extensions, calls are taped in a
common mailbox assigned to this feature. Extension users with access to the mailbox can retrieve and
listen to the recorded conversations. This parameter requires you to define the Mailbox for Call Taping.
You may select the Extension Type—SLT, DKP, SIP—whose mailbox you want to use for Call Taping from
the drop down list, and enter the Port Number of this extension. For example, if you want to select the
mailbox of the phone connected to SLT Extension Port 4, enter 4 in this field.
• Enable Programming through Comm. Port: You can configure ETERNITY via the Serial COM port of a
computer, using communication software like HyperTerminal, ProComm, and BitComm. To configure the
system using COM Port, you must enable this check box.
• Communication Port for Programming: If you have enabled Programming through Comm. Port,
select the COM Port to which you have connected the computer you will use for system configuration.
ETERNITY ME supports two COM Ports, you may select Comm. 1 or Comm. 2.
If your system is ETERNITY GE 6S, GE12S, PE6S or PE3SP which have a single COM Port, select
Comm. 1. For more information, see “Configuring using Serial COM Port”.
• BITE Card Slot Number: BITE (Built-In Test Equipment) is an automatic testing facility offered by the
ETERNITY to test the functioning of the SLT, Magneto, TWT, Loop Dial and E&M ports using BITE. For
more details, see “BITE (Built-In Test Equipment)”.
The “The SLT+MAG+TWT+LD+ENM Card” of ETERNITY supports BITE. If you have this card installed in
your system, you must define the BITE Card Slot, i.e. the slot in which the “The SLT+MAG+TWT+LD+ENM
Card” is installed.
• Enable Watch Dog: ETERNITY supports the Watchdog function to detect and restart the system,
whenever the system hangs. To use this function, select this check box. Default: Disabled.
When Watch Dog function is disabled, you must manually restart the system when it hangs.
• Master Buzzer: ETERNITY ME has a buzzer on the master card which sounds whenever the system
detects a fault. To know more, see “System Fault Log”.
You can set the duration of time for which the buzzer should sound by setting the Master Buzzer On Time
and OFF Time as Master Buzzer On Timer and Master Buzzer Off Timer respectively.
• Enable Incoming CLI Modification: To apply Incoming CLI modification, select this check box.
• Country Code: This is the Country Code of the country where ETERNITY is installed. The Country
Code helps ETERNITY detect whether the Incoming CLI received is a national or an international
number. Do not enter any prefix for the Country Code. For example, if your ETERNITY is installed in
USA, enter only ‘1’ as the Country Code. Do not enter ‘+’ or “00’ as prefix to the country code ‘1’.
Default: ‘91’ (India).
• International Prefix: These are digits required as Prefix for dialing International Numbers. The prefix
may be up to 5 digits, with numbers from 00000 to 99999. Default: ‘00’.
• National Prefix: These are digits required as Prefix for dialing long distance, National (within the
country) numbers. The prefix may be up to 5 digits, with numbers from 00000 to 99999. Default: ‘0’.
• Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose:
• No (Area Code not required), if your public telephone network does not require the dialing of Area
Code for local numbers.
• Yes (Area Code is required), if your public telephone network requires you to dial the Area Code for
local numbers.
• Yes, with Prefix Digit, if your public telephone network requires you to dial Area Code with a
particular Prefix for local numbers. If you select this option, you must also enter the prefix digits for
the area code for local calls in the Prefix Area Code field.
Clock Synchronization
The ETERNITY supports four clock sources for Clock Synchronization for the T1E1 and BRI ports. To know more,
see “Clock Synchronization”.
You must select the Clock Source in the order of Priority from 1 to 4. By default, the Clock Source priority is
selected as follows:
• Clock Source - Priority 1 - T1E1 - 001
• Clock Source - Priority 2 - T1E1 - 002
• Clock Source - Priority 3 - T1E1 - 003
• Clock Source - Priority 4 - T1E1 - 004
Also set the Clock Synchronization Frequency as: 8KHz Derived, 8KHz, 2.048MHz, 1.54 MHz
By default the following messages are configured as DND Text messages, you may customize these messages,
according to your requirement.
1 Do Not Disturb
2 Unavailable
3 In Meeting
4 In Conference
5 Try on Mobile
6 On Vacation
7 On Business Trip
8 Out of Office
9 With a Guest
Publish Message
ETERNITY offers 10 different text Messages to Publish Presence, as listed in the table below. You can customize
message 6 to 9 to match your requirement.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
9 Out of Office
When you connect a Door Lock release device, you must set the Normal Contact Type of the DOP to Normally
Open. Default: Normally Open. See “Digital Output Port (DOP)”.
• Instigation Signal: Depending on the application for which you are using the DIP, select the appropriate
instigation signal for the DIP as High or Low state.
• 'High' state signifies that the DIP is normally open. DIP instigation signal should be set as 'High'
when the sensor connected to the DIP keeps the Loop open and closes it to signal an event.
• 'Low' state signifies that the DIP is normally closed. DIP instigation signal should be set as 'Low'
when the sensor connected to the DIP normally keeps the Loop closed and opens/breaks it to
signal an event.
• Minimum Instigation Time: This is the time for which the instigation signal from the sensor device
should remain present on the DIP to be recognized by the DIP as a genuine signal. The range of this
timer is from 01 to 99 seconds. By default the Minimum Instigation Time is set to 01 second. You may
set the 'Minimum Instigation Time' to the desired value.
To know more about the usage of the DIP, see “Digital Input Port (DIP)” and “Automated Control
Applications”.
Customer Name
• To enter Customer Name, dial:
• 5401-Customer Name-#*
• To select the type of music to be played when stations are kept on hold, dial:
• 3552-Code
Where,
Code is
1 for Voice Module 01
2 for Analog Input Port
Default: 1.
• To select the type of music to be played when trunks are kept on hold, dial:
• 3553-Code
Where,
Code is
1 for Voice Module 01
2 for Analog Input Port
Default: 1.
If all the extensions of the Routing Group you selected for Alarm Notification type are busy, the extension
user will be played MoH (MoH can be Voice Module 01 or through Analog Input Port).
• To enable/disable the Disconnect when Caller Doesn't Dial a Digit' flag, dial:
• 5338-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable.
• To enable/disable the Disconnect DID call when Dialed Number Busy, dial:
• 5336-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable.
• To enable/disable Disconnect DID Call, when Dialed Number does Not Reply, dial:
• 5337-Code
Where, Code is
0 for Disable.
• To enable/disable Beep when Call Taping and Conversation Recording starts, dial:
• 5332-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Enable
• To disable/enable Feature Tone in place of Dial Tone when Call Forward is set, dial:
• 5312-Feature Tone Flag
Where,
Feature Tone Flag is
0 for Disable
1 for Enable
Default: Enable
• To select a Language for SE, SA and Front Desk User Web Interface, dial:
• 5319-Language
Where,
Language is
1 for English
2 for French
3 for German
4 for Spanish
5 for Portuguese
• To program minimum number of digits received in CLI to consider as call from the Public Network, dial:
• 5314-Minimum Caller ID Digits
Where,
Minimum Caller ID Digits is from 01 to 16.
Default:8
• To set the ON/OFF time of the ETERNITY ME Master Card buzzer, dial:
• 5308-On Timer-Off Timer
Where,
ON Timer and OFF Timer are 0000 to 9999 milliseconds.
Program the Country Code without any prefix. For example, if you want to program USA as Country
Code, enter ‘1’ only (without the prefix ‘00’ or ‘+’)
Program the Area Code without any prefix. For example, if you want to program Mumbo as Area Code,
enter ‘22’ only (without the prefix ‘0’)
Clock Synchronization
• To select clock source, dial:
• 5341-Clock Source Index-Port Type-Port Offset
Where,
Clock Source Index is 1 to 4.’
Port Offset is 01 to 32.
Clock Source Index is from 1 to 4.
05 T1E1 1-8
04 BRI 01-32
00 Null 000
Default:
1 T1E1-1
2 T1E1-2
3 T1E1-3
4 T1E1-4
1 8 KHz Derived
2 8 KHz
3 2.048 MHz
4 1.54 MHz
Default: 2.048 MHz for India and other countries except USA. For USA: 1.54 MHz.
1 Do Not Disturb
2 Unavailable
3 In Meeting
4 In Conference
5 Try on Mobile
6 On Vacation
On Business
7
Trip
8 Out of Office
9 With a Guest
If the new message has less than characters 16 terminate the string with #*.
For example: to program 'Out for Lunch' as the second DND Text Message, dial 1502-2- Out for
Lunch -#*
Publish Message
You can customize Publish Messages using Jeeves only.
• Exit SE mode.
What’s this?
Several features of the ETERNITY are based on Timers and Counts. For example, how long and how many times
an extension should ring when Message Wait is set, or how long the Busy Tone, the Ring Back Tone, or the Error
Tone should be played to an extension. ETERNITY allows you to configure most of these Timers and Counts to suit
your requirement. Listed below are the Timers and Counts related to the various features and facilities, along with a
brief description and default value of each.
Auto Redial
Auto Redial - Dial Tone The time for which the system waits to sense the 0 to 255 3 seconds
Wait Timer (sec.) dial tone from the PSTN/CO Network.
Auto Redial - Ring Back The time for which system waits to sense the 0 to 255 60 seconds
Tone Wait Timer (sec.) Ring Back Tone from the PSTN/CO Network
after dialing the requested number.
Auto Redial - Ring The time for which the extension that has 0 to 255 45 seconds
Timer (sec.) requested Auto Redial should ring.
Auto Redial - Normal The time interval between auto redial attempts 0 to 255 45 seconds
Timer (sec.) when Auto Redial ‘Normal’ is set.
Auto Redial - Normal The number of auto redial attempts the system will 0 to 255 5 tries
Count make when Auto Redial ‘Normal’ is set.
Auto Redial - Priority The time interval between auto redial attempts 0 to 255 10 seconds
Timer (sec.) when Auto Redial ‘Priority’ is set.
Auto Redial - Priority The number of auto redial attempts the system will 0 to 255 20 attempts
Count make when an extension having the feature Auto
Redial Priority in its Class of Service uses Auto
Redial ‘Priority’.
Dial Tone Timer (sec.) The time for which the system plays the Dial tone. 2 to 255 7 seconds
Ring Back Tone Timer The time for which the system plays the Ring Back 1 to 255 45 seconds
(sec.) Tone.
Busy Tone Timer (sec.) The time for which the system plays the Busy Tone. 1 to 255 7 seconds
Error Tone Timer (sec.) The time for which the system plays the Error Tone. 1 to 255 30 seconds
Feature Confirmation The time for which the system plays the 1 to 255 7 seconds
Tone Timer (sec.) Confirmation Tone when a feature is set or
canceled.
Programming Tone The time for which the system plays the 2 to 255 15 seconds
Timer (sec.) Programming Tone when you successfully enter the
SE mode from a phone.
Programming Error The time for which the system plays the Error Tone 1 to 255 3 seconds
Tone Timer (sec.) when you have entered an invalid command string
while configuring a feature from a phone.
Programming The time for which the system plays the 1 to 255 3 seconds
Confirmation Tone Confirmation Tone when a system command is
Timer (sec.) successfully executed when configuring the system
from a phone.
Tone Demo Timer (sec.) The time for which the system plays Call Progress 1 to 255 30 seconds
Tone when you are demonstrating the tone.
DID Inactivity Timer The time after which the system releases the 0 to 255 60 seconds
(sec.) trunk, if the caller has not dialed any digit, or
when a DID or Trunk Auto Answer call is not
answered by the landing destination.
DID Answer Wait Timer The time for which the system waits before 0 to 255 5 seconds
(sec.) answering a DID call.
DID Music Timer (sec.) The time for which the system plays music after 0 to 255 5seconds
answering the DID call.
DID Beeps Timer (sec.) The time for which the system plays beeps to the 0 to 255 10 seconds
caller to prompt the caller to dial the desired
extension number.
DID Ring Timer (sec.) The time for which the system rings on the 0 to 255 30 seconds
landing destination extension in a DID call.
DID Busy Tone Timer In a DID call, the time for which the system plays 0 to 255 15 seconds
(sec.) Busy Tone, if the dialed extension is busy.
DID Error Tone Timer In a DID call, the time for which the system plays 0 to 255 5 seconds
(sec.) Error Tone to the caller, if the caller has dialed an
invalid code.
DISA Idle State Timer In a DISA PIN Authentication call, the time for which 0 to 255 20 seconds
(sec.) the system waits for the caller to go Off-hook after
entering DISA. If the caller does not go Off-hook
within this timer, the system releases the call.
DISA Inactivity Timer In a DISA call, the time for which the system waits 0 to 255 2 minutes
(min.) for the caller to dial digits. If the caller does not dial
any digit within this timer, the system disconnects
the call. This timer is applicable only for Analog
Trunks.
Other Features
Auto Call Back Ring The time for which the extension requesting the 1 to 255 30 seconds
Timer (sec.) Auto Call Back and the destination extension will
ring.
Interrupt Request Timer The time for which the extension on which the 1 to 255 45 seconds
(sec.) Interrupt Request is made will get the beeps.
Barge-In Timer (sec.) The time after which the extension that has 1 to 255 10 seconds
activated Barge-In gets connected to the
extension which is barged in.
Trunk Reservation The time for which a trunk remains reserved for 1 to 255 10 minutes
Timer (min.) the extension that has reserved the trunk.
Transfer while Ringing When an extension transfers a call to another 1 to 255 30 seconds
Timer (sec.) extension after it starts ringing, this is the time for
which the system will wait for the transfer target
extension to answer the call. If the transfer target
does not answer the call within this timer, the call
is returned to the transferror.
Transfer on Busy Timer When a call is transferred to a Busy extension, 1 to 255 30 seconds
(sec.) this is the time for which beeps are played on the
transfer target extension.
Trunk to Trunk Inactivity In a Trunk-to-Trunk call, this is the time for which 1 to 255 2 seconds
Timer (sec.) the system waits after call maturity for any digit to
be dialed. If no digit is dialed within this timer, the
system drops the call.
Call Park Timer (sec.) The time after which the call comes back to the 2 to 255 45 seconds
extension that has parked the call.
Call Park Release Timer The time after which the parked call gets 1 to 255 3 minutes
(min.) disconnected.
Live Call Screening The time for which the speaker of the DKP/ 1 to 255 10 seconds
(sec.) Extended IP Phone extension remains ON while
the message from the caller is being recorded.
Message Wait Ring It is the Number of times the extension should 0 to 255 10 attempts
Count ring after the Message Wait is set on an
extension. This count is applicable only when
‘Ring’ is selected as the Message Wait
Notification type for the extension.
Message Wait Ring The time for which the extension rings to indicate 1 to 255 30 seconds
Timer (sec.) that Message Wait is set for the extension. This
timer is applicable only when ‘Ring’ is selected as
the Message Wait Notification type for the
extension.
Message Wait Ring The time after which the extension should ring 1 to 255 30 minutes
Interval Timer (min.) again to indicate Message Wait is set. This timer
is applicable only when ‘Ring’ is selected as the
Message Wait Notification type for the extension.
Conflict Dialing Timer The time for which the system waits for the 1 to 255 2 seconds
(sec.) extension user to dial the next digit to resolve
conflicting access codes dialed by the extension
user.
Station - Inter Digit Wait The time for which the system waits for the 2 to 255 7 seconds
Timer (sec.) extension user to dial the next digit. On the expiry of
this timer, the system considers it as the end of
number dialing.
Trunk - First Digit Wait the time for which the system waits for the 1 to 255 25 seconds
Timer (sec.) extension user to dial the first digit, after grabbing
the trunk.
Trunk - Inter Digit Wait When an extension user has grabbed the trunk and 1 to 255 3 seconds
Timer (sec.) is dialing a number, the system waits for the Trunk-
Inter-Digit wait timer for the extension user to dial
the next digit. On the expiry of this timer, the system
considers it as end of number dialing and proceeds
with the call.
Call Hold Retrieval This is the time for which a call put on Global 1 to 999 60 seconds
Timer (sec.) Hold remains connected in the system. If the call
put on Global Hold is not retrieved within this
timer, the call is returned to the DKP/Extended IP
Phone which put it on hold.
RCOC Record Delete This is the time for which the record of the 1 to 999 999 seconds
Timer (min.) outgoing call is stored in the RCOC Table. The
timer is activated whenever a record is stored in
the RCOC table. At the end of this timer, the
system deletes this record from the table.
Release Conference if This is the time for which the system will wait for 1 to 255 2 minutes
Idle for more than (min.) participants of a Dial-In Conference to withdraw or
release themselves from the conference, one-by-
one. On the expiry of this timer, the system releases
the Dial-In Conference and frees the resource
occupied by this conference in the conferencing
circuit.
Watchdog Refresh Wait This is the time within which the system should 1 to 255 60 seconds
Timer (sec.) signal to the Watchdog. If the system does not send
signal to the Watchdog within this Timer, the
watchdog device will restart ETERNITY.
The Timers and Counts on this page are arranged by the name of the feature or function these are related
to.
• Change the value of the Timer or Count by entering the desired duration or count in the respective fields.
Auto Redial
To set Auto Redial Normal - Timer, dial:
• 1704-Seconds
Where,
Seconds is from 000 to 255.
201. Time for which the system demonstrates the tone/ring to the user.
DISA
To set DISA Idle State Timer, dial:
• 2420-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 020 seconds.
Other Features
To set Auto Call Back Ring Timer, dial:
• 3801-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 030 seconds.
• Exit SE mode.
What’s this?
Access to the Eternity is secured at three levels by way of a password:
• at the System Engineer Level with the System Engineer (SE) password.
• at System Administrator Level with the System Administrator (SA) password.
• at the User Level with the User Password.
The System Engineer and the System Administrator passwords secure the system settings from access and
alteration by unauthorized persons (anyone other than the System Engineer and the System Administrator), thus
preventing possible misuse of the features and facilities.
• The SE password is stored in the Master Card/CPU Card. If you forget the SE password, the only way
to restore the default SE password is to change the Jumper settings of the Master Card/CPU card.
• You are advised to record and store the SE password at a safe place, where it can be accessed by you
(the System Engineer) to avoid the inconvenience of restoring the default SE password.
ETERNITY ME
ETERNITY GE
ETERNITY PE
The default SE password will be restored to 1234. You can now enter the programming mode by dialing 1#91-
1234 (the default password). You can also change the SE password again using Jeeves or by dialing a
command as described above.
The System Engineer can assign a new SA password from Jeeves or by dialing a command from a telephone.
If an extension user forgets the User Password, a new password can be issued to the extension user by the
System Administrator using Jeeves or by dialing a command.
a. locking the Keypad the extension phone; possible only on DKP extensions.
b. setting the User Status for the extension as "Absent"; possible on both DKP and SLT extensions. Read the
feature description “User Absent/Present” to know more.
The System Administration can do this lock the keypad of DKP extensions and set DKP and SLT extension users
as ‘Absent’ using Jeeves and by dialing commands from a telephone.
• Extension users can also set their status as 'Absent' or 'Present' from their respective extension
phones. Refer “User Absent/Present”).
• DKP extension users can also lock the keypad of their phones from the DKP Phone Menu. Refer
“Digital Key Phone-Operation” for instructions on navigating the phone menu.
• It is also possible to lock/unlock the DKP keypad and set the user extension status as 'Absent'/'Present'
from a remote location using “Direct Inward System Access (DISA)”.
What’s this?
• The T1 system format logs which are useful for maintenance purpose of the PBX, consist of:
• The ETERNITY supports T1-Statistics and Alarm-logs. This chapter explains both Statistics and Alarms
and ‘Loop Back Tests’ for T1.
• The T1 Maintenance consists of Error Counts (Performance Statistics), Alarms and Loop Back Test. This
is as per standards like G.704, G.706 and G.732. G.775 is also considered for detection of defect
conditions like LOS, LOF, AIS, etc. (Loss of Signal, Loss of Frames, Alarm Indication Signal).
• Digital line can have transmission errors. All the errors will not generate an Alarm. Few severe errors will
generate Alarms. However, all the errors will be logged in the System Fault Log.
• The SNIIC (subscriber Network interface integrated circuit) is used to interface T1 line to ETERNITY. It
supports error counters listed in the table given below. Each error detected by the ETERNITY ME Card
T1E1PRI/port will be sent to the system in form of an event.
• The system will count these errors and prepare a statistical record if the condition matches.
For Example:
Severely Errored Seconds Count is incremented when one OOF event reaches the master or more than 320
framing errors reach the master. This statistical record is updated and maintained by the master.
Performance Statistics
Error Counters supported by SNIIC
• Framing Bit Error Counter: This counter is incremented on receipt of any error in the framing pattern. In
D4, FS errors are counted. (FS errors are counted if enabled). In ESF, any error in the 001011 framing
pattern increments this counter.
• Out of Frame Counter: Out of Frame is the occurrence of a particular density of framing error events. For
D4 framing, OOF is declared when the receiver detects two or more framing errors within 0.75ms or two or
more errors out of five or fewer consecutive framing bits. It ends when there are fewer than two frame bit
errors within 0.75ms period. For ECF framing, OOF is declared when the receiver detects two or more
errors out of five or fewer consecutive framing bits. It ends when there are fewer than two frame bit errors
within 3ms period.
• CRC-6 Error Counter: This counter is incremented when the received frame has CRC-6 errors. This is
applicable for ECF framing only.
• Line Code Violation Error Counter: This counter is incremented when a bipolar violation error occurs or
when excessive zeroes event occurs.
• Positive Slip Counter: This counter is incremented every time a positive slip occurs.
• Negative Slip Counter: This counter is incremented every time a negative slip occurs.
Errored Seconds
Unavailable Seconds
Errored Seconds
For D4, it is defined as a second with one of the following:
For ESF signals-It is a count of one second interval with one of the following:
Slip defects are not counted in SES. Also this is not incremented during an Unavailable second. It is defined
as a second in which either an OOF has occurred or 320 or more framing errors have occurred.
Unavailable Seconds
• It is defined as a second for which T1 service is unavailable. An unavailable state is declared at the onset
of 10 consecutive severely errored seconds and is cleared on onset of 10 consecutive seconds with no
severely errored seconds.
How to configure
T1 FDL
T1 FDL can be enabled/disabled. This parameter is applicable only if Framing = ESF. If the Network (Public or
Private) to which the ETERNITY is connected does not support FDL then T1 FDL will be disabled.
T1 FDL Meaning
0 Disable
1 Enable
T1 FDL Protocol
ETERNITY will support both the protocols of reporting the performance monitoring. This parameter is applicable
only if T1 FDL is enabled and Framing = ESF. This parameter will match the protocol expected by the other end of
the link.
Use following command to program the T1 FDL Protocol for a T1E1PRI port:
6165-1-T1E1PRI-T1 FDL Protocol
0 Disable
1 AT&T 54016
2 ANSI T1.403
ANSI T1.403
• As per this standard, the receiving equipment transmits a performance report message (PRM) each
second over the FDL. This PRM is not sent to any specific remote location, but is broadcast so that any
PRM receiving device on the T1 line can intercept the message.
• The PRM contains error information pertaining to only the previous 4 seconds.
• It is the responsibility of the PRM receiver to accumulate the information and store it for 24 hours or the
time desired. This method allows performance monitoring points at different locations along the T1
network so that error localization is determined.
AT&T 54016
• As per this standard, the receiving equipment collects the data but does not transmit it on its own based on
time as done by ANSI. Instead, the transmitting end sends a request to the receiving end to transmit the
performance data.
Explanation of Alarms
• Alarms are indicated on the LEDs of the ETERNITY ME Card T1E1PRI. T1E1PRI card has four LEDs viz.
L1 to L4. L1 and L2 indicate alarms for T1E1PRI-1 whereas L3 and L4 indicate alarms for T1E1PRI-2.
• During normal conditions, the LED blinks green (1 sec. ON, 1 sec. OFF).
RED Alarm
• This alarm is generated if Loss of Signal persists for 2.5 seconds. This is indicated by flashing the LED
Red (500ms ON, 500ms OFF). The master logs this event in the System Fault Log. It is logged as System
Fault event 11 as RED Alarm <Slot No.> <Port No.> at HH:MM:SS.
It is cleared:
• When the signal is acquired back and persists for 10 seconds. The LED is turned OFF. The system logs
this event in the System Fault Log. For example, it is logged as a System Fault event viz. Fault index 12 as
RED Alarm Cleared <Slot No.> <Port No.> at HH:MM:SS.
• The interface SNIIC has settings for 32 consecutive zeroes or 192 consecutive zeroes. Hard program it to
32 consecutive zeroes.
• When RED Alarm is declared, Yellow Alarm is sent to the far end within 12ms of detection of LOS.
YELLOW Alarm
• This Alarm is also known as Remote Alarm Indication. This Alarm is generated when Yellow Alarm is sent
by the far end (Yellow Alarm is sent by the far end to indicate that it has lost the incoming signal).
It is declared:
• When the signal corresponding to Yellow Alarm persists for 0.5 seconds. This is indicated by flashing the
LED Orange (500ms ON, 500ms OFF). The system logs this event in the System Fault Log. For example
it is logged as a System Fault event 13 as YELLOW Alarm <Slot No.> <Port No.> at HH:MM:SS.
It is cleared:
• When No Yellow Alarm signal persists for 0.5 seconds. The LED is turned OFF. The master logs this event
in the System Fault Log. For example it is logged as a System Fault event 14 as YELLOW Alarm Cleared
<Slot No.> <Port No.> at HH:MM: SS.
Remarks:
Yellow Alarm in D4 is declared if:
• More than 285 zeroes are received in bit position 2 of incoming DS0 channels during an integration period
of 1.5ms.
• More than 3 ones are detected in bit position 2 of incoming DS0 channels during an integration period of
1.5ms.
• Yellow Alarm signal pattern 0000000011111111 does not occur in 10 contiguous 16-bit signal pattern
intervals.
• When clearance of AIS is detected for continuous 10 seconds. The LED is turned OFF. The master logs
this event in the System Fault Log. For example it is logged as a System Fault event viz. Fault index 10 as
BLUE Alarm Cleared <Slot No.> <Port No.> at HH:MM:SS.
• If less than six zeroes are received on the incoming line data during a 3 ms interval. AIS is cleared if the
above condition does not exist for 3 ms. This interval of 3 ms could be upto maximum of 75ms.
When BLUE Alarm is declared, Yellow Alarm is sent to the far end.
LED Indication
• LED L1 and L2 are assigned to port 1 and LED L3 and L4 are assigned to port 2.
• LED L1 is used for Card Heart Bit as well as status of the PORT1.
LED L1 is used for both Port 1 status and System Heart Bit. So in case LED is flashing, it will flash for 1
second and will OFF for 1 second.
Relevant Topic:
1. “T1E1 Trunks” 1608
What’s this?
Digital Signal Level 1 (T1E1) trunks use Bit-Oriented Signaling (BOS) and multiplexes 24 channels (T1 service) or
32 channels (E1 service) into a single data stream. T1E1 can be used for voice or voice-grade data and for data-
transmission protocols. T1 trunk service multiplexes 24 channels into a single 1.544-Mbps data stream. E1 trunk
service multiplexes 32 channels into a single 2.048-Mbps stream. Both T1 and E1 provide a digital interface for
trunk groups.
Signaling Modes:
Common Channel Signaling (CCS) is an industry-standard technique where any one of a group of channels carries
the signals for the other channels. Matrix uses the 24th channel of a group for signaling. This signaling technique
differs from 24-channel signaling. When the system is configured for Facility-Associated Signaling, 24-channel
signaling uses the 24th channel in a T1E1 facility to carry signals. This technique also is called clear channel, out-
of-band or alternate voice data (AVD) signaling.
Channel Associated Signaling (CAS) is similar to common-channel signaling and is used only when the Bit Rate is
2.048 Mbps (the trunk is used with an E1 interface). Signaling is carried on the 16th channel.
Common-channel signaling and channel associated signaling provide a maximum transmission rate of 64 Kbps for
bearer channels.
ROBBED-BIT signaling is a per-channel signaling technique for transmitting signaling bits on each channel in a
T1E1 facility. The least-significant bit in every 6th transmitted information frame is removed and replaced by a
signaling bit. This technique is also called in-band signaling. The maximum transmission rate for each bearer
channel with ROBBED-BIT signaling is 56 Kbps.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps connection and on the 16th channel for a 2.048
Mbps connection.
QSIG is an ISDN based protocol for signaling between nodes of a Private Integrated Services network.
Any of the common trunks, except for PCOL (Personal Central Office Line) trunks, can be analog or digital. (PCOL
trunks can only be analog.) Administering a digital trunk group is very similar to administering its analog
counterpart, but digital trunks must connect to a T1E1 port and this port must be administered separately.
• In case of ISDN_E1_PRI and ISDN_E1_CAS the protocol supports 32 Channels ranging from 00 to 31,
out of which 2 channels (channel no. 00 and 16) are used for framing/signaling. So effectively user has 30
channels for OG/IC calls.
• For better understanding of the user the channel IDs are mapped as shown below. Thus for the E1_PRI
and E1_CAS the T1E1 portsupports total 30 channels ranging from 01 to 30, which you can use for
making and receiving calls.
Channel ID as
00 01 02 03 04 14 15 16 17 18 31
per Protocol
Channel ID for
01 02 03 04 14 15 16 17 30
User Interface
Mapping of Actual Channel ID with User Interface (E1_PRI and E1_CAS)
• Similarly, in case of T1 PRI, Protocol supports 24 channels (from 01 to 24), in which channel no. 24 is used
for the signaling, so effectively there are 23 Voice channels are available.
• But in case of T1 RBS, Protocol supports 24 channels (from 01 to 24) and the protocol doesn’t consume
any channel for signaling so that there are total 24 channels available for the users.
• The ETERNITY will form the line loop back or payload loop back same as required by the other end.
• Far end Loop Back Test: These are of two type viz.
• These tests are conducted when the T1E1 port is in NT mode and is connected to other PBX and wants to
test the line between itself and the far end. In this mode, ETERNITY acts as a network.
• When the other end connected with the T1E1 port of the ETERNITY wants to perform the loopback tests,
the T1E1 port will form the loopback depending upon the type of test the other end wants to perform (i.e.
line loopback or payload loopback).
• The protocol doesn't support the facility that the remote end can close/open the loop at the T1E1
port side automatically.
• Hence when the remote end wants to perform the loopback test he will inform the person (SE) at
the ETERNITY side to form the type of loop back as desired (i.e. Line loopback or Payload
loopback) on the T1E1 port.
• When the SE forms the loopback at T1E1 port side (by issuing appropriate SE Command), the
remote end can start the test.
• On completion of the testing, the remote end person will inform the SE to release/open the loopback
formed at the T1E1 port side.
• On request of the remote end, the SE will give SE command for the T1E1 port to open the Near end
loopback.
Case 2:
• The protocol supports loopback Activation and deactivation message, whereby the remote end can
send the loop up activation code to the T1E1 port and the T1E1 port decodes the message and
forms the loop back automatically.
• On completion of the testing Remote end can send the loop deactivation code and the T1E1 port
can open the already formed loopback.
• So in this case the SE's intervention is not required to form and release the loop back.
• In case the remote end doesn't support facility to automatically form/release the loopback for the
T1E1 port though the carrier is T1, the SE can use the command (6141-1-T1E1-Type of Loop Back)
on request of the remote end and the loopback will get formed or released, depending on the
command issued.
Loopback Activation:
• When the system receives the SE command to form Line/Payload loop back (in case of E1):
• It will inform the ETERNITY ME Card T1E1 about the received command ETERNITY ME Card T1E1
will release all the calls supported by the T1E1 Port under test.
• The ETERNITY ME Card T1E1 will form the required type of loop back.
• System will put the T1E1 port in maintenance mode. It will release all active calls supported by T1E1
port and restrict the usage of T1E1 port for IC/OG calls.
• ETERNITY ME Card T1E1 will release all the calls supported by the T1E1 Port under test.
• System will put the T1E1 port in maintenance mode. It will release all active calls supported by T1E1 port
and restrict the usage of T1E1 port for IC/OG calls.
Loopback Release:
• On receiving the loopback release code or receiving the command to release the loop back (either
Payload or Line), the system will take the T1E1 port out from the maintenance mode and now the T1E1
port will function normally.
• Line loopback
• Payload loopback
The option 'Release All Loop Backs' is required when the far end requires the universal loop back release
code to release the Loop Back.
Case 1:
• The protocol doesn't support the facility that the ETERNITY can perform the loopback tests
automatically.
• The SE of the ETERNITY will inform the far end connected with the T1E1 port, to form the loopback
as required.
• Once the far end has formed the desired loop back the SE can issue command to start the Far end
loop back test.
• On receiving the command to start Far End loop back test (either the Line Loopback or Payload
loopback test).
• ETERNITY ME Card T1E1 will release all the calls supported by the T1E1 Port under test.
• The ETERNITY ME Card T1E1 will start the PRBS generator and counter.
• ETERNITY ME Card T1E1 will send the PRBS count every 1 second to the system.
• ETERNITY ME Card T1E1 will increment PRBS Counter for every error encountered during the test
during one second.
• ETERNITY ME Card T1E1 will reset the PRBS counter to zero, after sending the PRBS Counter to
the system, every second.
• The system will store the received PRBS count (received every second) in Performance report,
which can be captured from Serial port or Printer port.
• When the SE wants to end the loopback test he will issue the command to end the Far End Loop
Back Test, for the T1E1 port under the test.
• SE will inform the other end's person that loopback test is finished and now the remote end can
open the loop formed.
• On receiving the command to end loop back test for the T1E1 port, the system will take the T1E1
port out of maintenance state and inform the T1E1 port about the received command.
Case 2:
• When the line type of the T1E1 port is configured as "ISDN_T1_PRI" or "ISDN_T1_RBS"
• The activation methods are different for the D4 and ESF framing.
• In case of D4 framing only Line loop back is supported.
• In case of ESF framing both, Line loopback and Payload loop back are supported.
When Framing = D4
D4 Framing supports only:
• The system will inform the ETERNITY ME Card T1E1 about the command and put this T1E1 port in
maintenance state.
• The ETERNITY ME Card T1E1 will send the line loop back "Activation Code" and will start the PRBS
generator and counter.
• ETERNITY ME Card T1E1 will send the PRBS count every 1 second to Master.
• ETERNITY ME Card T1E1 will increment PRBS Counter for every error encountered during the test during
one second.
• ETERNITY ME Card T1E1 will reset the PRBS counter to zero, after sending the PRBS Counter to the
system, every second.
• The system will store the received PRBS count (received every second) in Performance report, which can
be captured from Serial port or Printer port.
• The system will inform the ETERNITY ME Card T1E1 about the received command and will take out the
T1E1 port from the maintenance mode.
• The system will put the T1E1 port under test in Maintenance mode and will inform the T1E1 Port about the
received command.
• ETERNITY ME Card T1E1 will release all the calls supported by the T1E1 port.
• ETERNITY ME Card T1E1 will send the Loop back "Activation Message" for the line/payload loop back as
informed by the system.
• The ETERNITY ME Card T1E1 will start the PRBS generator and counter.
• ETERNITY ME Card T1E1 will send the PRBS count every 1 second to Master.
• ETERNITY ME Card T1E1 will increment PRBS Counter for every error encountered during the test during
one second.
• ETERNITY ME Card T1E1 will reset the PRBS counter to zero, after sending the PRBS Counter to Master,
every second.
• The system will inform the ETERNITY ME Card T1E1 about the received command and will take out the
T1E1 port from the maintenance mode.
• ETERNITY ME Card T1E1 will send the "Deactivation Message" for the line/payload loopback test as
required.
Performance Report:
• 50 entries will be stored in Performance report.
• The report will store the entries in FIFO order.
From SA Mode
User the following command to start/abort online printing of T1E1 performance report:
1072-030-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Flag is 0.
User the following command to start/abort offline printing of T1E1 performance report:
1072-031-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Flag is 0.
• During loop back test the PRBS counter may be greater than zero at initial stage of the loop back
stage, but it will be zero afterwards consistently for the healthy condition.
• PRBS counter = greater than zero, indicates the 'faulty' condition for the loop back test.
Port Status
Used to enable/disable the port. When the Port is disabled, it will not be allotted to the user on grabbing the port.
Instead the user will get error tone.
0 Disable
1 Enable
Service Provider
Line coding is a pattern that data assumes as it is propagated over a communications channel as following:
• AMI: Alternate Mark Inversion. All transmissions generated by digital trunk ports are encoded in AMI Line
coding. Voltage on the line will be a net DC '0'.
• For both B8ZS and HDB3, line coding, the terminals at end-to-end will be HDB3-compatible.
1 AMI-Basic
2 B8ZS
3 HDB3
Set Line Coding = AMI or B8ZS for T1 line. However, B8ZS is recommended.
Set Line Coding = AMI or HDB3 for E1 line. However, HDB3 is recommended.
CMI is used in Japan and since ETERNITY does not support Japan, this option will never be used.
Framing Mode
• Framing: It is the set of 24 or 32 8-bit time slots that is treated as a single transmission unit.
• SF: D4 Superframe. The 12-frame unit that contains the synchronization pattern is known as the
'Superframe'.
• ESF: Extended Superframe Unlike the 12-bit synchronization pattern for D4 which utilizes all of the
available framing bits for sync., ESF employs only 6 of the available 24 framing bits to carry a
synchronization pattern. Each 24-frame entity spanning one ESF cycle, is referred to as an 'ESF
superframe'. With CEPT1 framing, the framing information is cycled through sixteen (0-15) frames (each
containing channels 0-31). The type of framing used at both ends must be identical, unless the signaling
and framing are converted somewhere in the transmission stream (as might happen in international
communication).
Use following command to program the Framing Mode for the T1E1 port:
6104-1-T1E1-Framing
6104-2-T1E1-T1E1-Framing
6104-*-Framing
Where,
Framing Meaning
2 ESF
Set Framing = SF or ESF for T1 line. However, ESF is recommended since it supports advanced features
like CRC and FDL, which provide the performance reports.
Signal Type/Line Type signifies the type of signaling to be used on the T1/E1 line. The E&M Protocol can be used
on T1E1 Port. This can be done by assigning E&M Feature Template to T1E1 Port. Refer chapter “E&M Feature
Template”.
1 ISDN_E1_PRI
2 ISDN_T1_PRI
3 ISDN_E1_CAS
4 ISDN_T1_RBS
5 ISDN_E1_QSIG
6 ISDN_T1_QSIG
7 ISDN_E1_E&M
8 ISDN_T1_E&M
DDI Routing is not supported on T1/E1 trunk line if you have selected E&M as the Signal Type.
Interface Companding
The entry in this field must match the compounding method used by the far-end switch. This field does not appear
for all T1E1 ports. Companding is a method of improving the signal-to-noise (S/N) ratio resulting from the pulse
code modulation (PCM) process on voice calls. The analog signal's amplitude is compressed before it is quantized
and transmitted. Either of two algorithms are used to compand voice band signals: A-law and m-law. A-law is
generally used in countries that use E1 at 2.048 mbps; while m-law is used in countries that use T1 at 1.544 mbps.
Use the following command to program interface companding (PCM coding) of a T1E1:
6108-1-T1E1-Interface Companding
6108-2-T1E1-T1E1-Interface Companding
6108-*-Interface Companding
Where,
T1E1 is from 1 to 8.
Mode Meaning
0 Disable
1 Enable
This field increases the strength of incoming signals by a fixed amount to compensate for line losses.
Use the following command to program the receive equalization parameters of a T1E1:
6111-1-T1E1-Receive Equalization Parameters
6111-2-T1E1-T1E1-Receive Equalization Parameters
6111-*-Receive Equalization Parameters
Where,
T1E1 is from 1 to 8.
1 None
2 8 dB
3 16 dB
4 24 dB
5 32 dB
6 40 dB
7 48 dB
8 Reserve
Glare Option
ISDN glare occurs if the system initiates an out going call on a B-Channel at the same time the network initiates an
incoming call on that same B-channel. When processing a glare condition, programmed glare option on T1E1 port
will be considered.
Use following command to program Glare Option for the T1E1 port:
1 Proceed
2 Held Back
Idle Code
It is the 8-bit sequence that occupies the time slot on a E1/T1 trunk channel when it is not being used.
The binary equivalent of the programmed value (000 to 255) is sent on the channel to signify that the channel is
idle. (or Unused) This setting depends on the network. Most commonly applicable values are 7F and FF (Binary
equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Use message mode of the Digital Switch IC to send the idle channel code.
This timer is relevant while receiving the called party number information in overlap receiving mode.
If overlap receiving timer value is 0, it means indirectly to support only Enbloc receiving method. By the term
'Enbloc', we mean receiving digits in a single block.
This Timer is used to provide delay for the number dialing by the T1E1 port. Pause timer will be applicable when
any 'P' digit is configured in the DTMF number string which is to be outdialed as DTMF digits on T1E1 port. One of
the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage Dialing”.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY will out dial
the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
DTMF ON Time
This parameter decides for how much time the DTMF digit will be ON, while out dialed by the ETERNITY.
One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage Dialing”.
This parameter decides how much time the pause (gap) should be present between two digits while dialed by the
ETERNITY. One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage
Dialing”.
Orientation Type
Use following command to program 'Orientation Type' for the T1E1 port:
6106-1-T1E1-Orientation Type
6106-2.T1E1-T1E1-Orientation Type
6106-*-Orientation Type
Where,
Orientation Meaning
1 Terminal
2 Network
3 Tie Line
By default Type = 1.
When Orientation = Terminal, the port will be regarded as trunk. All the trunk related parameters will be applicable.
When Orientation = Network, the port will be regarded as station. All the station related parameters will be
applicable.
When Orientation = Tie-line, the port will be regarded as station for all IC calls to it and as trunk for all OG calls to
be made through it.
Use following command to program OG source calling party TON for a T1E1:
6126-1-T1E1-Source TON
6126-2-T1E1-T1E1-Source TON
6126-*-Source TON
Where,
T1E1 is from 1 to 8.
Unknown: This is used when the user or network has no a prior information about
1 the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
2 International Number.
6 Abbreviated Number
7 Reserved Number.
Use following command to program OG source calling party NPI for T1E1:
6127-1-T1E1-Source NPI
6127-2-T1E1-T1E1-Source NPI
6127-*-Source NPI
Where,
1 Unknown
4 Telex Numbering
Use following command to program OG Destination Called Party TON for a T1E1:
6128-1-T1E1-Destination TON
6128-2-T1E1-T1E1-Destination TON
6128-*-Destination TON
Where,
T1E1 is from 1 to 8.
1 Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
2 International Number.
6 Abbreviated Number
7 Reserved Number.
Use following command to program OG Destination Called Party NPI for T1E1:
6129-1-T1E1-Destination NPI
6129-2-T1E1-T1E1-Destination NPI
6129-*-Destination NPI
Where,
1 Unknown
4 Telex Numbering
Use following command to program to select whether the inband tones should be feed on T1E1-NT before sending
DISCONNECT message?
6130-1-T1E1-Flag
6130-2-T1E1-T1E1-Flag
6130-*-Flag
Where,
Flag Meaning
0 No
1 Yes
Default = NO.
When this flag is enabled (set as 'Yes'), inband tones shall be feed for 15 seconds (fixed, non programmable)
before sending DISCONNECT message.
When this flag is disabled (set as 'No'), inband tones (Busy/Error as applicable for the state of the call) shall not be
feed before sending the DISCONNECT message. However when DISCONNECT message is sent from T1E1-NT
port, inband tones will always be sent with 'progress indicator 8'.
• When the user dials the trunk access code or selective trunk access code for dialing the number directly
on the trunk port, he waits for the dial tone before dialing the number. But some exchange does not give
Dial Tone for the T1E1 Port. For Example, when T1E1 port as E1CAS type is used in Delhi, it is observed
that the exchange does not give dial tone when direct dialing on the trunk is used.
• To solve such problems, ETERNITY supports 'Dial Tone' when the T1E1 Port is accessed.
• It is applicable only when Online dialing is used as for Store and Forward dialing, the dial tone is given to
the user.
Use following command to program the dial tone flag for T1E1 port:
6115-1-T1E1-Flag
6115-2-T1E1-T1E1-Flag
6115-*-Flag
Where,
T1E1 is from 1 to 8.
Flag Meaning
0 Disable
1 Enable
By default, Dial Tone Flag is '0' for all the T1E1 ports.
• When dial tone flag is disabled, user will hear the dial tone of the exchange if provided, otherwise, user
will hear the silence.
• If the user is making the call from the FXS port and dial tone is not provided by the exchange, user will
not know when to start dialing the number. In this case, it is possible that some digits are not out dialed
on the port and wrong number is dialed out because system will out dial the number only if Outgoing
call Acknowledge is received from the ETERNITY ME Card T1E1 and user will not know about this
condition. Hence, it is required to enable this flag, if exchange is not providing the dial tone.
• When online dialing or store and forward dialing is used, some exchange do not provide any tone when
call is proceeding from the exchange.
• Thus, the user does not know whether the call is processing or not if for some time there is silence from the
exchange.
• Routing tone will be stopped when alert message or connect message or disconnect message comes from
the ETERNITY ME Card T1E1.
Use following command to program the routing tone flag for T1E1 port:
6116-1-T1E1-Flag
6116-2-T1E1-T1E1-Flag
6116-*-Flag
Where,
T1E1 is from 1 to 8.
Flag Meaning
0 Disable
1 Enable
By default, Routing Tone Flag is '0' for all the T1E1 ports.
Use the following command to program number of channels reserved for OG channel count:
6136-1-T1E1-Channel Count (OG)
6136-2-T1E1-T1E1-Channel Count (OG)
6136-*-Channel Count (OG)
Where,
T1E1 is from 1 to 8.
Channel Count (OG) is from 00 to 30. "It specifies the number of channels to be reserved for making an OG calls.
For example, If OG channel count is programmed as 15, simultaneous 15 (maximum) OG calls can be made from
the T1E1 port".
By default, OG Channel Count is 30.
Use the following command to program number of channels reserved for IC channel count:
6137-1-T1E1-Channel Count (IC)
6137-2-T1E1-T1E1-Channel Count (IC)
6137-*-Channel Count (IC)
Where,
T1E1 is from 1 to 8.
Channel Count (IC) is from 00 to 30. "It specifies the number of channels to be reserved for making an IC calls. For
example, If IC channel count is programmed as 10, simultaneous 10 (max.) IC calls can be received on the T1E1
port".
By default, Channel Count (IC) is 30.
OG Reference ID
Use the following command to assign IC Route Reference ID for Non-working Hour:
6134-1-T1E1-IC Reference ID
6134-2-T1E1-T1E1-IC Reference ID
6134-*-IC Reference ID
Where,
T1E1 is from 1 to 8.
IC Reference ID is from 00 to 99.
By default, IC Route ID is 00.
Use the following command to activate/release Near End Loopback for T1E1:
6141-1-T1E1-Loopback
Where,
T1E1 is from 1 to 8.
Loopback Meaning
Use the following command to start/stop far end loopback test for T1E1:
6142-1-T1E1-Code
Where,
T1E1 is from 1 to 8.
Code Meaning
Use following command to assign the port for online Performance Report Printing:
6143-Port
Where,
Port Meaning
0 None
1 COM 1
2 COM 2
3 Printer
Default, None.
Use following command to assign the port for offline Performance Report Printing:
6144-Port
Where,
Port Meaning
0 None
1 COM 1
2 COM 2
3 Printer
Default, None.
Use following command to program E1 Line Signaling Variant for the T1E1 port:
6152-1-T1E1-E1 Line Signaling Variant
6152-2-T1E1-T1E1-E1 Line Signaling Variant
6152-*-E1 Line Signaling Variant
Where,
T1E1 is from 1 to 8.
01 ITU T Q.400-Q.490
• ITU-T Q.400-Q.490 is the only Line Signaling Variant supported currently. India supports three types of E1
line signaling protocols. One is same as ITU-T Q.400-Q.490. Other two differ from this standard and hence
are not supported currently. These will be supported on knowing their usage in the field. Most of the
Countries follow ITU-T Q.400-Q.490.
Use following command to program E1 Register Signaling Variant for the T1E1 port:
6153-1-T1E1-E1 Register Signaling Variant
6153-2-T1E1-T1E1-E1 Register Signaling Variant
6153-*-E1 Register Signaling Variant
Where,
T1E1 is from 1 to 8.
1 E1 CAS Decadic
2 E1 CAS DTMF
3 E1 CAS MFC R2
4 E1 CAS MFC R1
• E1 CAS Decadic: A-bit is used to transmit the DNIS. The A-bit is toggled as per the Pulse Dial Ratio set.
• E1 CAS DTMF: DNIS/ANI is transmitted in the corresponding speech channel using the DTMF signals as
per ITU-T Q.23.
• E1 CAS MFC R2: DNIS/ANI is transmitted in the corresponding speech channel using the MFC R2 signals
as per ITU-T Q.400-Q490.
• E1 CAS MFC R1: DNIS/ANI is transmitted in the corresponding speech channel using the MFC R1 signals
as per ITU-T Q.300-Q390. Generally, E1 CAS MFC R2 is only used.
E1 Auto Alarm
E1 Auto Alarm-This command is used to disable/enable E1 Auto Alarm for T1E1 port.
Flag Meaning
This field reduces the outgoing signal strength by a fixed amount. The appropriate level of loss depends on the
distance between your switch (measured by cable length from the smart jack) and the nearest repeater. Where
another switch is at the end of the circuit, as in campus environments, use the cable length between the 2 switches
to select the appropriate setting from the table below. This field is relevant if the Near-end CSU type field is
integrated.
Use the following command to program the line build out parameters of a T1E1:
6162-1-T1E1-Code
6162-2-T1E1-T1E1-Code
6162-*-Code
Where,
T1E1 is from 1 to 8.
Code Meaning
1 0-133ft
2 133-266ft
3 266-399ft
4 399-533ft
5 533-665ft
6 -7.5dB or equidistance
7 -16dB or equidistance
8 -22.5dB or equidistance
Use following command to enable/disable Custom Pulse Width (CPW) Flag for the T1E1 port for T1 signaling:
6171-1-T1E1-Flag
6171-2-T1E1-T1E1-Flag
6171-*-Flag
Where,
T1E1 is from 1 to 8.
Flag Meaning
0 Disable
1 Enable
Use following command to program Custom Pulse Width Word 1 for the T1E1 port for T1 signaling:
6172-1-T1E1-Custom Pulse Width Word 1
6172-2-T1E1-T1E1-Custom Pulse Width Word 1
6172-*-Custom Pulse Width Word 1
Where,
T1E1 is from 1 to 8.
Custom Pulse Width Word 1 is from 000 to 127 volt.
By default, Custom Pulse Width Word 1 is 63 volt.
Use following command to program Custom Pulse Width Word 2 for the T1E1 port for T1 signaling:
6173-1-T1E1-Custom Pulse Width Word 2
6173-2-T1E1-T1E1-Custom Pulse Width Word 2
6173-*-Custom Pulse Width Word 2
Where,
T1E1 is from 1 to 8.
Custom Pulse Width Word 2 is from 000 to 127 volt.
By default, Custom Pulse Width Word 2 is 58 volt.
Use following command to program Custom Pulse Width Word 3 for the T1E1 port for T1 signaling:
6174-1-T1E1-Custom Pulse Width Word 3
Use following command to program Custom Pulse Width Word 4 for the T1E1 port for T1 signaling:
6175-1-T1E1-Custom Pulse Width Word 4
6175-2-T1E1-T1E1-Custom Pulse Width Word 4
6175-*-Custom Pulse Width Word 4
Where,
T1E1 is from 1 to 8.
Custom Pulse Width Word 4 is from 000 to 127 volt.
By default, Custom Pulse Width Word 4 is 00 volt.
Use following command to enable/disable Custom Pulse Width (CPW) Flag for the T1E1 port for E1 signaling:
6155-1-T1E1-Flag
6155-2-T1E1-T1E1-Flag
6155-*-Flag
Where,
T1E1 is from 1 to 8.
Flag Meaning
0 Disable
1 Enable
Use following command to program Custom Pulse Width Word 1 for the T1E1 port for E1 signaling:
6156-1-T1E1-Custom Pulse Width Word 1
6156-2-T1E1-T1E1-Custom Pulse Width Word 1
6156-*-Custom Pulse Width Word 1
Where,
T1E1 is from 1 to 8.
Custom Pulse Width Word 1 is from 000 to 127 volt.
By default, Custom Pulse Width Word 1 is 109 volt.
Use following command to program Custom Pulse Width Word 2 for the T1E1 port for E1 signaling:
Use following command to program Custom Pulse Width Word 3 for the T1E1 port for E1 signaling:
6158-1-T1E1-Custom Pulse Width Word 3
6158-2-T1E1-T1E1-Custom Pulse Width Word 3
6158-*-Custom Pulse Width Word 3
Where,
T1E1 is from 1 to 8.
Custom Pulse Width Word 3 is from 000 to 127 volt.
By default, Custom Pulse Width Word 3 is 064 volt.
Use following command to program Custom Pulse Width Word 4 for the T1E1 port for E1 signaling:
6159-1-T1E1-Custom Pulse Width Word 4
6159-2-T1E1-T1E1-Custom Pulse Width Word 4
6159-*-Custom Pulse Width Word 4
Where,
T1E1 is from 1 to 8.
Custom Pulse Width Word 4 is from 000 to 127 volt.
By default, Custom Pulse Width Word 4 is 064 volt.
This field selects the variant for ISDN PRI when ISDN PRI is selected as signaling mode. This must match with the
other end of the link.
1 ATT_4ESS
2 ATT_SESS
3 AUSTRALIA
4 DMS
5 NET5
6 NTT_INS64
7 SWV_HONG_KONG
8 US_NI12
This timer signifies the maximum time for which the forward signal can be ON, from the outbound end.
• The DSP (Digital Signaling Processor) device will send the forward tone for this timer and will expect
backward signal within this timer. If no backward signal is received during this time, a timeout condition will
occur in which case, an alert signal will be sent to the ETERNITY ME Card Master, error tone be issued to
the calling party and a clear forward signal will be sent on the line.
This timer signifies the maximum time between two out going forward signals. During this time the forward tone will
remain OFF. If the outbound end does not send a forward signal for this time, the inbound end will interpret it as per
its condition and shall take action accordingly.
Use the following command to program Forward Tone Maximum Off Timer:
7102-1-T1E1-Forward Tone Maximum OFF Timer
7102-2-T1E1-T1E1-Forward Tone Maximum OFF Timer
7102-*-Forward Tone Maximum OFF Timer
Where,
T1E1 is from 1 to 8.
Forward Tone Maximum OFF Timer is from 01 to 99 seconds.
By default, Forward Tone Maximum Off Timer is 24 seconds. This timer will be less than the Incoming R2 register
timeout timer programmed by the inbound end.
This timer signifies the maximum time within which one Compelled signalling cycle should end.
Use the following command to program the Maximum compelled cycle time:
7103-1-T1E1-Maximum Compelled Cycle Time
7103-2-T1E1-T1E1-Maximum Compelled Cycle Time
Backward signals A-3, A-4, A-6 and A-15 are pulsed to the outbound end. Pulse duration of these signals vary from
country to country.
Use the following command to program Pulse duration for pulsed signals:
7104-1-T1E1-Pulse Duration for Pulsed Signals
7104-2-T1E1-T1E1-Pulse Duration for Pulsed Signals
7104-*-Pulse Duration for Pulsed Signals
Where,
T1E1 is from 1 to 8.
Tolerance is fixed at +/-25ms.
By default, the Pulse Duration for Pulsed Signals is 150ms.
This timer signifies the time for which the outbound end waits for the pulsed signal. If the pulsed signal is not
received during this time, the compelling signalling is considered to be complete.
Use the following command to program the Pulsed Signal Maximum Wait Timer:
7105-1-T1E1-Pulsed Signal Maximum Wait Timer
7105-2-T1E1-T1E1-Pulsed Signal Maximum Wait Timer
7105-*-Pulsed Signal Maximum Wait Timer
Where,
T1E1 is from 1 to 8.
Pulsed Signal Maximum Wait Timer is from 01 to 99 seconds.
By default, Pulsed Signal Maximum Wait Timer is 15 secs.
This timer signifies the minimum time between receipt of line seizure signal and first forward signal.
Use the following command to program first forward tone wait timer:
7106-1-T1E1-First Forward Tone Wait Timer
7106-2-T1E1-T1E1-First Forward Tone Wait Timer
7106-*-First Forward Tone Wait Timer
Where,
T1E1 is from 1 to 8.
First forward tone wait timer is from 08 to 24 seconds.
By default, the First Forward Tone Wait Timer is 15 seconds.
This timer signifies the minimum time for which the forward/backward signal should persist on the line to be
recognized as a forward/backward signal by the receiving end.
This parameter is applicable only when the DNIS length is set= 99 (i.e variable). The outbound end indicates end of
DNIS using a group I tone or using time out.
Use following command to program to set DNIS END Type (outbound) for T1E1 port:
7108-1-T1E1-End of DNIS
7108-2-T1E1-T1E1-End of DNIS
7108-*-End of DNIS
Where,
T1E1 is from 1 to 8.
End of DNIS is from 00, 11 to 15.
00 indicates End of DNIS as time out.
01 to 15 indicates group 1 tone to declare End of DNIS.
By default, DNIS End Type (Outbound) is 15.
This parameter is applicable only when the DNIS length is set = 99 (i.e variable). The inbound end indicates end of
DNIS using a Group I tone or using time out.
This parameter signifies the number of DNIS digits after which to send ANI. As such ANI is sent on receiving the
backward tone 'Send next digit' or send next ANI digit. If send next ANI tone is received then this parameter is not
applicable. But if same tone is used by the inbound end to request next ANI digit and next DNIS digit then ANI is
sent after the number of digits as set in this parameter.
This parameter indicates the Group A tone (received from the inbound tone) that should be interpreted as a
question by the inbound end asking the outbound end whether the outbound end has ANI digits to be sent.
This parameter signifies the Group 1 tone that the outbound end will send to the inbound end as a response to Is
ANI Available tone from the inbound end. The tone defined in this parameters indicates the Group 1 tone with
which the Outbound end will respond to the inbound end to indicate that it has ANI digits to be sent.
Use the following command to program the Positive Response to Is ANI Available (Outbound):
7112-1-T1E1-Positive Response to Is ANI Available
7112-2-T1E1-T1E1-Positive Response to Is ANI Available
7112-*-Positive Response to Is ANI Available
Where,
T1E1 is from 1 to 8.
Positive Response to Is ANI Available is a Group 1 tone from 01 to 15.
By default, Positive Response to Is ANI Available is 01.
This parameter signifies the Group 1 tone that the outbound end will send to the inbound end as a response to Is
ANI Available tone from the inbound end. The tone defined in this parameter indicates the Group 1 tone with which
the Outbound end will respond to the inbound end to indicate that it does not have ANI digits to be sent.
Use following command to program the Negative Response to Is ANI Available (Outbound):
7113-1-T1E1-Negative Response to Is ANI Available
7113-2-T1E1-T1E1-Negative Response to Is ANI Available
7113-*-Negative Response to Is ANI Available
Where,
T1E1 is from 1 to 8.
Negative Response to Is ANI Available is a Group 1 tone from 01 to 15.
By default, Negative Response to Is ANI Available is 10.
This parameter signifies the Group 1 tone used to signify end of ANI digits with Presentation Allowed.
Use the following command to program the End of ANI with Presentation Allowed (Outbound):
7114-1-T1E1-ANI End Tone with Presentation Allowed (Outbound)
7114-2-T1E1-T1E1-ANI End Tone with Presentation Allowed (Outbound)
7114-*-ANI End Tone with Presentation Allowed (Outbound)
Where,
T1E1 is from 1 to 8.
This parameter signifies the Group 1 tone used to signify end of ANI digits with Presentation Restricted.
Use the following command to program the End of ANI with Presentation Restricted (Outbound):
7115-1-T1E1-ANI End Tone with Presentation Restrict (Outbound)
7115-2-T1E1-T1E1-ANI End Tone with Presentation Restrict (Outbound)
7115-*-ANI End Tone with Presentation Restrict (Outbound)
Where,
T1E1 is from 1 to 8.
ANI End Tone with Presentation Restrict (Outbound) is a Group 1 tone from 00, 11 to 15. If no such tone is sent, set
this parameter to 00.
By default, ANI End Tone with Presentation Restrict (Outbound) is 00.
This parameter specifies the number of DNIS digits that are expected by the Inbound end to indicate the Called
party number during MFC R2 signaling.
• DNIS Digit length (01 to 98) will be expected by the inbound end. (Practical value would be 01 to 10)
• DNIS Digit length 99 indicates DNIS length is variable. Further action is taken after timeout or on receipt of
I-15. Refer parameter 'DNIS End Type (Inbound)'.
The inbound end may request/may not request ANI digits. It may request ANI digits after receiving first DNIS or
after receiving second DNIS or even after receiving all the DNIS digits.
Use following command to program the number of DNIS digits after which ANI digits should be requested by the
inbound end:
7117-1-T1E1-ANI Request Position
7117-2-T1E1-T1E1-ANI Request Position
7117-*-ANI Request Position
Where,
T1E1 is from 1 to 8.
ANI Request Position is from 00, 01 to 99.
ANI Request Position=00 indicates Never request ANI digits.
ANI Request Position=01 to 98 indicates Request ANI digits on receipt of these many DNIS digits.
ANI Request Position = 99 indicates Request after receiving all the DNIS digits (complete DNIS).
By default, ANI Request Position is 99.
This parameter signifies the number of ANI digits that would be expected by the inbound side as Calling Party
Number during MFC R2 signaling. This parameter at the inbound side guides the inbound register to switch from
requesting ANI digits back to requesting DID digits.
• ANI Length = 99 indicates ANI Length is variable. If ANI length is variable, the logic waits for End of ANI
from the outbound side. The inbound end will sense for I-12 and I-15. I-12 is used to signify that no ANI
digits are available whereas I-15 is used to signify end of ANI digits. Some countries like China use I-15 to
signify both the events viz. End of ANI and no ANI digits available.
This parameter specifies the backward group A tone used to ask the outbound end whether it has ANI digits to be
sent. This parameter is also known as Request ANI Category.
This parameter specifies the Group 1 forward tone to be received by the inbound end from the outbound which
would indicate that Outbound end has ANI digits to be sent. This parameter is also known as ANI category.
• This cannot be zero. This is because; Is ANI Available request would be made by the inbound end only if
the country supports this protocol. In such event, Is ANI Available request will be responded to.
For example, In India I-1 or I-10 is sent by the Outbound end. In Kuwait, I-6 is sent.
This parameter specifies the Group 1 forward tone to be received by the inbound end from the outbound which
would indicate that Outbound end has ANI digits to be sent. This parameter is also known as ANI category.
• This cannot be zero. This is because; Is ANI Available request would be made by the inbound end only if
the country supports this protocol. In such event, Is ANI Available request will be responded to.
For example, In India I-1 or I-10 is sent by the Outbound end. In Kuwait, I-6 is sent.
This parameter specifies the Group I tone that the inbound end should expect from the outbound end to consider
End of ANI digits with an information that the Presentation of ANI by the outbound end is allowed.
Use following command to program the ANI End Tone Presentation Allowed:
7122-1-T1E1-ANI End Tone Presentation Allowed
7122-2-T1E1-T1E1-ANI End Tone Presentation Allowed
7122-*-ANI End Tone Presentation Allowed
Where,
T1E1 is from 1 to 8.
By default, ANI End Tone Presentation Allowed is 15.
• ANI End Tone Presentation Allowed is 00 or 11 to 15. If no such tone is sent, set this parameter to 00. For
For example, India uses A-4, China uses A-1, etc.
This parameter specifies the Group I tone that the inbound end should expect from the outbound end to consider
End of ANI digits with an information that the Presentation of ANI by the outbound end is Restricted.
Use following command to program the ANI End Tone Presentation Restricted:
7123-1-T1E1-ANI End Tone Presentation Restricted
7123-2-T1E1-T1E1-ANI End Tone Presentation Restricted
7123-*-ANI End Tone Presentation Restricted
Where,
T1E1 is from 1 to 8.
By default, ANI End Tone Presentation Restricted (Inbound) is 00.
• ANI End Tone Presentation Restricted is 00 or 11 to 15. If no such tone is sent, set this parameter to 00.
For example: India uses A-4, China uses A-1, etc.
Ordinary Subscriber
Priority Subscriber
This parameter specifies the forward group II tone used to inform the inbound end that the calling party is a Priority
Subscriber. This signal is sent in response to Calling Party Category signal Request from the inbound end.
Maintenance Equipment
This parameter specifies the forward group II tone used to inform the inbound end that the calling party is
maintenance equipment.
Operator
This parameter specifies the forward group II tone used to inform the inbound end that the calling party is Operator.
Pay Phone
Data Transmission
This parameter specifies the forward group II tone used to inform the inbound end that the call is a Data Call.
Interception Operator
This parameter specifies the forward group II tone used to inform the inbound end that the call is from Interception
Operator. This is used in Singapore.
This parameter specifies the backward group A tone used to request next digit, be it ANI digit or DNIS digit.
• For example: India uses A-1 to signify event 'Send DNIS Digit'.
This parameter specifies the backward group A tone used to request last but one digit i.e. N-1 digit, be it ANI digit or
DNIS digit.
Use following command to program the Send last but one Digit:
7132-1-T1E1-Send Last But One Digit
7132-2-T1E1-T1E1-Send Last But One Digit
7132-*-Send Last But One Digit
Where,
T1E1 is from 1 to 8.
Send Last But One Digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Last But One Digit (N-1) is 02.
• For example: India uses A-9 to signify event 'Send last but one digit'.
This parameter specifies the backward group A tone used to request last but two digit i.e. N-2 digit. Be it ANI digit or
DNIS digit.
Use following command to program the Send last but two digit:
7134-1-T1E1-Send Last But Two Digit
7134-2-T1E1-T1E1-Send Last But Two Digit
7134-*-Send Last But Two Digit
Where,
T1E1 is from 1 to 8.
Send Last But Two Digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Last But Two Digit (N-2) is 07.
• For example: India uses A-7 to signify event 'Send last but two digit'.
This parameter specifies the backward group A tone used to request last but three digit i.e. N-3 digit. Be it ANI digit
or DNIS digit.
Use following command to program the Send Last But Three Digit:
7135-1-T1E1-Send Last But Three Digit
7135-2-T1E1-T1E1-Send Last But Three Digit
7135-*-Send Last But Three Digit
Where,
T1E1 is from 1 to 8.
Send last but three digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Last But Three Digit is 08.
• For example: India uses A-8 to signify event 'Send last but three digit'.
This parameter specifies the backward group A tone used to inform the inbound end that the incoming register at
the inbound end needs no additional address digit and is about to go over to transmission of a group B signal
conveying the status of the equipment at the subscriber at the inbound end.
Use following command to program the Address-Complete, Change over to reception of Group B signals:
7136-1-T1E1-Address-Complete, Change Over to Reception of Group B Signals
This parameter specifies the backward group A tone used by the inbound end to request Calling Party Category
from the outbound end. This tone also informs the outbound end to change to reception of Group C signal. This
signal is used in Mexico.
Use following command to program the Send Calling Party Category and Change to Group C:
7137-1-T1E1-Send Calling Party Category and Change to Group C
7137-2-T1E1-T1E1-Send Calling Party Category and Change to Group C
7137-*-Send Calling Party Category and Change to Group C
Where,
T1E1 is from 1 to 8.
Send Calling Party Category and change to Group C is from 00, 01 to 15. 00 is used for No tone.
By default, Send Calling Party Category and Change to Group C is 00.
This parameter specifies the backward group A tone used to inform the congestion at the inbound end.
This parameter specifies the backward group A tone used to request calling party category.
• For example: India uses A-7 to signify event 'Send last but two digit'.
This parameter specifies the backward group A tone used to inform the inbound end that the incoming register at
the inbound end needs no additional address digit, but will not send Group B signals. Also charge the call on
answer.
This parameter specifies the backward group A tone used to inform the outbound end to send all the DNIS digits
from the beginning.
Use following command to program the repeat DNIS digits from beginning of Group B signals:
7141-1-T1E1-Repeat DNIS Digits from Beginning
7141-2-T1E1-T1E1-Repeat DNIS Digits from Beginning
7141-*-Repeat DNIS Digits from Beginning
Where,
T1E1 is from 1 to 8.
Repeat DNIS Digits from Beginning is from 00, 01 to 15. 00 is used for No tone.
By default, Repeat DNIS Digits from Beginning is 00.
This parameter specifies the backward group A tone used to request next (first) ANI digit.
• Few countries use different tone to request next ANI digit and next DNIS digits.
This parameter specifies the backward group B tone used to inform the outbound end that the call cannot be made
through because of reasons beyond those which are considered by the Protocol and hence Special Information
tone will be sent to the calling party. The PBX should send only the Group B signal and then disconnect the call.
This parameter specifies the backward group B tone used to inform the outbound end that the call cannot be made
through because of reasons beyond those which are considered by the Protocol and hence Special information
tone will be sent to the calling party and request the outbound end to setup speech conditions. In this case, the PBX
will connect the calling party to the voice message of the PBX informing the caller that the call cannot be
connected. This is required in countries like Argentina. This is same as a condition 'Call rejected, No indication of
cause'. This signal is also used in India to inform the caller that the called party's number is changed and the caller
should contact the Help desk of the Service Provider.
Use following command to program the Send Special Information Tone and Setup Speech Condition:
7144-1-T1E1-Send Special Information Tone and Setup Speech Conditions
7144-2-T1E1-T1E1-Send Special Information Tone and Setup Speech Conditions
7144-*-Send Special Information Tone and Setup Speech Conditions
Where,
T1E1 is from 1 to 8.
Send Special Information Tone, and Setup Speech Conditions is from 00, 01 to 15. 00 is used for No tone.
By default, Send Special Information Tone and Setup Speech Conditions is 02.
This parameter specifies the backward group B tone used to inform the outbound end that the called subscriber is
busy.
This parameter specifies the backward group B tone used to inform the outbound end that the called subscriber is
free and the call is to be charged on answer.
This parameter specifies the backward group B tone used to inform the outbound end that the called subscriber is
free but the call is not to be charged on answer. This signal permits nonchargable calls without the need for
transferring "no charge' information by line signals.
Congestion
This parameter specifies the backward group A tone used to inform that congestion is encountered after
changeover from Group-A to Group-B signals.
Unallocated Number
This parameter specifies the backward group B tone used to inform the outbound end that the number received is
not in use.
This parameter specifies the backward group B tone used to inform the outbound end that the called subscriber's
line is out of order
This parameter specifies the Group B backward tone used to inform the outbound end that the call is rejected but
there is no indication of cause.
This parameter specifies the Group B backward tone used to inform the outbound end that the call is accepted and
the speech path is made through. This is same as A-6. This is required in few countries like Czech Republic.
This parameter specifies the Group B backward tone used to inform the outbound end that the number dialed by
the calling party is changed. However, this parameter will be rarely used. Please note that VM should be assigned
for this feature.
This parameter specifies the backward group C tone to request next (even first) ANI digit from the outbound end.
Use following command to program the Send next ANI digit (Group C):
7154-1-T1E1-Send Next ANI Digit (Group C)
7154-2-T1E1-T1E1-Send Next ANI Digit (Group C)
7154-*-Send Next ANI Digit (Group C)
Where,
T1E1 is from 1 to 8.
Send next ANI digit (Group C) is from 00, 01 to 15. Use '00' when this parameter is not applicable.
This parameter is applicable in Mexico only.
By default, Send Next ANI Digit (Group C) is 00.
This parameter specifies the backward group C tone to restart from first DNIS and request transition to Group A.
Use following command to program the Request transition to Group A and restart from first DNIS.
7155-1-T1E1-Request Transition to Group A and Restart from First DNIS
This parameter specifies the backward group C tone used to signify Address completed, change to reception of
Group B signal.
Use following command to program the Address completed, change to reception of Group B signal:
7156-1-T1E1-Address Completed, Change to Reception of Group B Signal
7156-2-T1E1-T1E1-Address Completed, Change to Reception of Group B Signal
7156-*-Address Completed, Change to Reception of Group B Signal
Where,
T1E1 is from 1 to 8.
Address Completed, Change to Reception of Group B Signal is from 00, 01 to 15. Use '00' when this parameter is
not applicable.
By default, Address Completed, Change to Reception of Group B Signal is 00.
Congestion (Group C)
This parameter specifies the backward group C tone used to signify Congestion
This parameter specifies the backward group C tone used to signify request transition back to group A, and send
next DNIS.
Use following command to program the tone for request transition back to group A, and send next DNIS signal:
7158-1-T1E1-Request Transition Back to Group A, and Send Next DNIS
7158-2-T1E1-T1E1-Request Transition Back to Group A, and Send Next DNIS
7158-*-Request Transition Back to Group A, and Send Next DNIS
Where,
T1E1 is from 1 to 8.
Request transition back to group A, and send next DNIS is from 00, 01 to 15. Use '00' when this parameter is not
applicable.
Request transition back to Group A, and restart the last DNIS (Group C)
This parameter specifies the backward group C tone used to signify request transition back to group A, and repeat
the last DNIS
Use following command to program the tone for request transition back to group A, and repeat the last DNIS:
7159-1-T1E1-Request Transition Back to Group A, and Restart the Last DNIS
7159-2-T1E1-T1E1-Request Transition Back to Group A, and Restart the Last DNIS
7159-*-Request Transition Back to Group A, and Restart the Last DNIS
Where,
T1E1 is from 1 to 8.
Request Transition Back to Group A, and Restart the Last DNIS is from 00, 01 to 15. Use '00' when this parameter
is not applicable.
By default, Request Transition Back to Group A and Restart the Last DNIS is 00.
CD Bits
This parameter indicates the default values of C and D bits when the ETERNITY transmits line signals.
0 00 (C=0, D=0)
1 01
2 10
3 11
By default, CD Bits is 1.
The C and D bits received during an IC call are ignored by the ETERNITY.
Invert Bit A
This parameter signifies that A-bit should be inverted before transmitting and on receiving.
Use following command to program to invert/don't invert Bit A for the T1E1 port:
7162-1-T1E1-Invert Bit A
7162-2-T1E1-T1E1-Invert Bit A
7162-*-Invert Bit A
Where,
0 Don't Invert
1 Invert
Invert Bit B
This parameter signifies that B-bit should be inverted before transmitting and on receiving.
Use following command to program to invert/don't invert Bit B for the T1E1 port:
7163-1-T1E1-Invert Bit B
7163-2-T1E1-T1E1-Invert Bit B
7163-*-Invert Bit B
Where,
T1E1 is from 1 to 8.
0 Don't Invert
1 Invert
Invert Bit C
This parameter signifies that C-bit should be inverted before transmitting and on receiving.
Use following command to program to invert/don't invert Bit C for the T1E1 port:
7164-1-T1E1-Invert Bit C
7164-2-T1E1-T1E1-Invert Bit C
7164-*-Invert Bit C
Where,
T1E1 is from 1 to 8.
0 Don't Invert
1 Invert
Invert Bit D
This parameter signifies that D-bit should be inverted before transmitting and on receiving.
Use following command to program to invert/don't invert Bit D for the T1E1 port:
7165-1-T1E1-Invert Bit D
7165-2-T1E1-T1E1-Invert Bit D
7165-*-Invert Bit D
Where,
0 Don't Invert
1 Invert
E1 Metering Bit
This parameter signifies the bit used by the network to signal metering pulses.
Use following command to program the E1 Metering Bit for the T1E1 port:
7166-1-T1E1-E1 Metering Bit
7166-2-T1E1-T1E1-E1 Metering Bit
7166-*-E1 Metering Bit
Where,
T1E1 is from 1 to 8.
0 None
1 Bit-A
2 Bit-B
3 Bit-C
4 Bit-D
This timer signifies the minimum time for which the metering bit should change to be recognized as a genuine
metering pulse subject to E1 Metering Pulse Maximum timer. All changes occurred for time less than this timer will
be ignored. This parameter is applicable only for E1 lines with R2 MFC signaling.
Use following command to program the Metering Pulse Minimum timer for the T1E1 port:
7167-1-T1E1-E1 Metering Pulse Minimum Timer
7167-2-T1E1-T1E1-E1 Metering Pulse Minimum Timer
7167-*-E1 Metering Pulse Minimum Timer
Where,
T1E1 is from 1 to 8.
E1 Metering Pulse Minimum timer is from 20ms to 1000ms.
By default, E1 Metering Pulse Minimum Timer is 150ms.
This parameter signifies the signal used to signify that the called party has disconnected the line first. This is
indicated in two ways viz. Release guard signal or forced release signal. This is country dependent.
Use following command to program the Clear Back Signal for the T1E1 port:
7168-1-T1E1-Clear Back Signal
7168-2-T1E1-T1E1-Clear Back Signal
Release Timer
This timer signifies the time for which the clear back signal should persist on the line to be recognized as a genuine
clear back signal. This is also known as Clear Back timer.
Use following command to program the Release Timer for the T1E1 port:
7169-1-T1E1-Release Timer
7169-2-T1E1-T1E1-Release Timer
7169-*-Release Timer
Where,
T1E1 is from 1 to 8.
Release Timer is from 20ms to 1000ms.
By default, Release Timer is 400 ms.
This timer signifies the time for which the outbound end waits for seizure acknowledgment from the inbound end
after sending the line seizure signal. On expiry of this timer, Clear forward signal is sent by the outbound end. Alarm
will be generated. This timer is applicable only when acting as Outbound end.
This timer signifies the time for which inbound register waits before declaring the channel idle (sending idle signal)
when clear forward line signal is received from the outbound end. This timer is also applicable for Forced Release
signal. This timer is applicable only when acting as Inbound end.
ETERNITY supports E&M protocol on T1E1 port. Both E1 E&M and T1 E&M can be used as options of signaling
type of T1E1 port. Select the signal type/line type as ISDN_E1_ E&M or ISDN_T1_ E&M.
1 ISDN_E1_PRI
2 ISDN_T1_PRI
3 ISDN_E1_CAS
4 ISDN_T1_RBS
5 ISDN_E1_QSIG
6 ISDN_T1_QSIG
7 ISDN_E1_E&M
8 ISDN_T1_E&M
Now proceed to program other parameters for E&M on T1E1 using following commands:
Use following command to select B Bit Value
7191-1-T1E1-Code
7191-2-T1E1-T1E1-Code
7191-*-Code
Where,
T1E1 is from 1 to 8.
Code Meaning
1 Same as A bit
2 Fixed Value
Use following command to program to invert/don't invert Bit A for the T1E1 port:
7162-1-T1E1-Invert Bit A
7162-2-T1E1-T1E1-Invert Bit A
7162-*-Invert Bit A
Where,
T1E1 is from 1 to 8.
0 Do not invert
1 Invert
Use following command to program to invert/don't invert Bit B for the T1E1 port:
7163-1-T1E1-Invert Bit B
7163-2-T1E1-T1E1-Invert Bit B
7163-*-Invert Bit B
Where,
T1E1 is from 1 to 8.
0 Do not invert
1 Invert
Use following command to program to invert/don't invert Bit C for the T1E1 port:
7164-1-T1E1-Invert Bit C
7164-2-T1E1-T1E1-Invert Bit C
7164-*-Invert Bit C
0 Do not invert
1 Invert
Use following command to program to invert/don't invert Bit D for the T1E1 port:
7165-1-T1E1-Invert Bit D
7165-2-T1E1-T1E1-Invert Bit D
7165-*-Invert Bit D
Where,
T1E1 is from 1 to 8.
0 Do not invert
1 Invert
Use the following command to program the line build out parameters of a T1E1:
6162-1-T1E1-Code
6162-2-T1E1-T1E1-Code
6162-*-Code
Where,
T1E1 is from 1 to 8.
Code Meaning
1 0 - 133 ft.
6 -7.5 dB or equidistance
7 -16 dB or equidistance
8 -22.5 dB or equidistance
Use following command to set flag for 'Gateway Application-Answer Signaling' on T1E1 trunk:
6119-1-T1E1-Gateway Application-Answer Signaling flag
6119-2-T1E1-T1E1-Gateway Application-Answer Signaling flag
6119-*- Gateway Application-Answer Signaling flag
Where,
T1E1 is from 1 to 8.
Flag Meaning
0 Disable
1 Enable
Use following command to program DTMF digits string to be dialed as Gateway Application-Answer Signaling:
6120-1-T1E1-Gateway Application-Answer Signaling DTMF String
6120-2-T1E1-T1E1-Gateway Application-Answer Signaling DTMF String
6120-1-T1E1-Gateway Application-Answer Signaling DTMF String
Where,
T1E1 is from 1 to 8.
DTMF Digits allowed for DTMF string are from (0 - 9), *, #, A, B, C, D.
Maximum 4 DTMF digits can be programmed. If you need less than 4 digits for DTMF string, terminate the
command using #*.
A #4
B #5
C #6
D #7
* **
# ##
0 Disable
1 Enable
1 DID
5 Operator
1 CLI Number
2 Alternate Number
Relevant Topics:
1. “Direct Dialing-In (DDI)” 1148
2. “DDI Routing Table” 1081
3. “T1 RBS Parameters” 1660
4. “T1 Maintenance” 1600
5. “E&M Feature Template” 565
6. “Multi-Stage Dialing” 1331
7. “Gateway Application-Answer Signaling” 1236
8. “Call Budget on Trunk” 874
9. “Logical Partition” 1313
Online report:
What’s this?
• Some countries like North America support the standard of 1.544Mbps of PCM trunk. This is known as T1
Trunks. The T1 type of PCM Trunks use Robbed Bit Signaling. ROBBED-BIT signaling is a per-channel
signaling technique for transmitting signaling bits on each channel in a T1E1 facility. The least-significant
bit in every 6th transmitted information frame is removed and replaced by a signaling bit. This technique is
also called in-band signaling. The maximum transmission rate for each bearer channel with ROBBED-BIT
signaling is 56 Kbps.
• ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps connection and on the 16th channel
for a 2.048 Mbps connection. There are two types of parameters:
• E&M Wink, E&M FGD, E&M Delay, E&M Immediate, FXS Ground start.
• FXS Loop Start, FXO Loop Start and FXO Ground Start.
• T1 Line signaling type is applicable when the Line Type is programmed as T1 RBS for the T1E1 Port.
T1 Line signaling is applicable when the Line Type is programmed as ISDN_T1_RBS for the T1E1 Port.
Refer chapter “T1E1 Trunks” to program Line Type as T1 RBS.
Making an OG Call
• To transmit Off-Hook, bits A, B, C and D on the transmit channel are set to 1.
• The far end (network) sends a wink (a momentary OFF-Hook for 200ms). i.e. bits A, B, C and D on the
receive channel receive a pulse (Active High) of 200ms.
• On receipt of the wink signal, DNIS (Dialed Number Identification Service or DID) is sent on the speech
channels using the Register Signaling type (DTMF or Decadic (A-Bit) or R1 MFC or R2 MFC).
DNIS is Dialed Number In Service. It is the ISDN number that is being dialed. This is provided by the telco in
the call setup messages. DNIS can be used to provide differentiated service to dialing users.
The call goes through when the called party answers the call.
• The far end sends the DNIS in the speech channels using Register Signaling.
Disconnect
• By the Network-Bits A and B on the receive channel are 0. Bits C and D are also 0 in ESF. Following this,
the bits on the transmit channel are set to 0 by the PBX.
• By the PBX-Bit A and B on the transmit channel are set to 0. Bit C and D are also set to 0 in case of ESF.
Following this, the bits A and B(C and D in ESF) are received as 0 on the receive channel.
• Bits A and B are set to 1 to indicate OFF-Hook. Bits A and B are set to 0 to indicate ON-Hook. Bits C and
D follow bits A and B (Incase of ESF).
Making an OG Call
• The far end (network) sends a wink (a momentary OFF-Hook for 200ms.). i.e. bits A, B, C and D on the
receive channel receive a pulse (Active High) of 200ms.
• On receipt of the wink signal, DNIS (Dialed Number Identification Service or DID. It is the ISDN number
that is being dialed. This is provided by the telco in the call setup messages. DNIS can be used to provide
differentiated service to dialing users.) is sent on the speech channels using the Register Signaling type.
• The call goes through when the called party answers the call.
Receiving an IC Call
• The far end sends the DNIS in the speech channels using DTMF.
• The PBX goes off hook when the call is answered by the called party.
Disconnect
• By the Network-Bits A and B on the receive channel are 0. Bits C and D are also 0 in ESF. Following this,
the bits on the transmit channel are set to 0 by the PBX.
• By the PBX-Bit A and B on the transmit channel are set to 0. Bit C and D are also set to 0 in case of ESF.
Following this, the bits A and (C and D in ESF) are received as 0 on the receive channel.
Please note that while using T1 RBS, only DID is being sent/received and not the CLI/ANI.
Direction State A B C D
Transmit ON-Hook 0 1 0 1
Receive ON-Hook 0 1 0 1
Receive OFF-Hook 0 1 0 1
Receive Ringing 1 1 1 1
Making an OG Call
• To transmit OFF-Hook, bits A (and Bit C if ESF) on the transmit channel is set to 1.
• The ETERNITY sends the DNIS (Dialed Number Identification Service or DID) on the speech channels
using the DTMF signals.
• The call goes through when the called party answers the call.
Receiving an IC Call
• The Network sends the DNIS on the speech channel using DTMF signals.
• The PBX detects toggling of bit B. When the called station of the PBX answers, the PBX transmits Off-
hook state by changing bit-A from 0 to 1.
Disconnect
By the Network:
• No indication from the Network. The PBX will detect error tone to detect a disconnect from the network. On
detecting on-hook from the network, the PBX transmits on-hook by setting Bit A from 1 to 0.
By the PBX:
Direction State A B C D
Receive Ringing 0 0 0 0
Making an OG Call
• To transmit Off-hook, bits A (and Bit C if ESF) and B (and Bit C if ESF) on the transmit channel is set to 0.
• The network detects this change and goes off-hook. The A-bit on the receive channel goes from 1 to 0.
The B-bit is set to 1.
• PBX detects dial tone and sends DNIS (DTMF digits) on the corresponding speech channel.
• The call goes through when the called party answers the call.
Receiving an IC Call
• The Network sends the DNIS on the speech channel using DTMF signals.
• When the called station of the PBX answers, the PBS transmits Off-hook state by changing bit-A from 0 to
1.
Disconnect
• By the Network-A bit on the receive channel of the PBX goes from 0 to 1. On detecting on-hook from the
network, the PBX transmits on-hook by setting Bit A from 1 to 0.
• FXO Loop Start or FXO Ground Start is used when the ETERNITY is connected to the Network i.e. when
the T1E1 port is configured for Terminal mode (Connection mode).
• Whereas FXS Loop Start or FXS Ground Start is used when the ETERNITY is connected to another
ETERNITY i.e. when the T1E1 port is configured for Network mode.
How to configure
Step 1
T1 Line Signaling Variants
Program the line Signaling Variant for the T1E1 Port. This command is applicable when the Bit Rate = T1. The T1
Line Signaling Variants are as per standard EIA-464B/AT&T TR41458. (This standard specifies the “Requirements
of the PBX Switching Systems”).
Use following command to program the T1 Line Signaling Variants for the T1E1 port:
6181-1-T1E1-T1 Line Signaling Variants
6181-2-T1E1-T1E1-T1 Line Signaling Variants
6181-*-T1 Line Signaling Variants
Where,
T1E1 is from 1 to 8.
Step 2
Wink Timer-This timer signifies the wink time. Wink is defined as a momentary off-hook for the time defined. It acts
as an acknowledgment signal to the end making an O/G call.
Step 3
Wink Wait Timer-This timer signifies the maximum time to wait before sending a wink start signal after an I/C
seizure is detected.
Use the following command to program the T1 wink wait timer for T1E1:
6183-1-T1E1-Wink Wait Timer
6183-2-T1E1-T1E1-Wink Wait Timer
6183-*-Wink Wait Timer
Where,
T1E1 is from 1 to 8.
Wink Wait Timer is from 0001 to 9999 ms.
By default, T1 Wink Wait Timer is 0200 ms.
Step 4
Wait Wink Timer-This timer signifies the time for which the system should wait for receiving the wink after sending
the OG Seizure.
Use the following command to program the T1 wait wink timer for T1E1:
6184-1-T1E1-Timer
6184-2-T1E1-T1E1-Timer
6184-*-Timer
Where,
T1E1 is from 1 to 8.
Timer is from 0001 to 9999 ms.
By default, T1 Wait Wink Timer is 0200 ms.
Step 5
Delay Duration-This duration signifies the time after which the DNIS information is to be sent (while making an OG
call)
Use the following command to program the T1 delay duration for T1E1:
6185-1-T1E1-Delay Duration
6185-2-T1E1-T1E1-Delay Duration
6185-*-Delay Duration
Where,
T1E1 is from 1 to 8.
Delay Duration is from 0001 to 9999 ms.
By default, T1 Delay Duration is 0140 ms.
Step 6
Start Delay Duration-This timer signifies the time for which the PBX waits for receiving DNIS from the network.
This timer is loaded on receiving the Off-hook (I/C Seizure) on the receive channel (while receiving an IC call)
Use the following command to program the T1 start delay duration for T1E1:
Step 7
DTMF Digit Timer-This timer signifies the DTMF digit time
Use the following command to program the T1 DTMF digit timer for T1E1:
6187-1-T1E1-DTMF Digit Timer
6187-2-T1E1-T1E1-DTMF Digit Timer
6187-*-DTMF Digit Timer
Where,
T1E1 is from 1 to 8.
DTMF Digit Timer is from 001 to 255 ms.
By default, T1 DTMF Digit Timer is 100 ms.
Step 8
DTMF Inter Digit Timer-This timer signifies the Inter-digit timer between the DTMF digits being dialed for the DNIS
(or the DID) information.
Use the following command to program the T1 DTMF inter digit timer for T1E1:
6188-1-T1E1-DTMF Inter Digit Timer
6188-2-T1E1-T1E1-DTMF Inter Digit Timer
6188-*-DTMF Inter Digit Timer
Where,
T1E1 is from 1 to 8.
DTMF Inter Digit Timer is from 001 to 255 ms.
By default, T1 DTMF Inter Digit Timer is 100 ms.
Step 9
Debug for T1E1 Port
ETERNITY supports debug of parameters (debug codes) depending on the Level of debug. On issuing this
command the ETERNITY ME Card T1E1 will send the debug details to the COM port of the T1E1 port.
Please note following command change as per Version of Software used. First set of Command '6191' is
used up to Version V6R0.12 and Commands at the end of 'Level 4' are used for Version V6R0.13
onwards. Here 'XXX' is the Code as mentioned in Tables for Level1 to level 4.
Option 1
Use following command to start/stop debug the parameters for the T1E1 port:
6191-1-T1E1-Level-Debug Code
Level 1:
001 CAS
002 MFC R2
008 Layer 4
Level 2:
Unused Unused Unused HDLC (D-Channel) FDL ABCD Bits Counters Alarms
001 Alarms
002 Counters
008 FDL
HDLC (D
016
Channel)
Level 3:
Unused Flow Debug NLS Debug LAP Debug SVC Primitives Variables State Primitives
001 Primitives
002 State
004 Variables
001 OS Task
002 NI Debug
Default: Debug Code = ‘Debug OFF’ for all T1E1 ports for all levels.
Option 2
Use following command to enabled Global Level-1 debug for T1E1 Port (ETERNITY-ME and ETERNITY-GE)
619-1-1-T1E1-1-Code
619-1-2-T1E1- T1E1-1-Code
619-1-*-1-Code
Where,
Code is from 000 to 255
Default is 000.
Option 3
Use following command to enabled Global Level-2 debug for T1E1 Port (ETERNITY-ME and ETERNITY-GE)
6191-1-T1E1-2-Code
6191-2-T1E1-T1E1-2-Code
6191-*-2-Code
Where,
Code is from 000 to 255.
Default is 000.
While using 6191 command:
• In ETERNITY GE, issue * Command only.
• In ETERNITY ME with ETERNITY ME Card T1E1 Dual, issue command for both the T1E1 software
port number assigned to T1E1 Dual card.
• In ETERNITY ME with ETERNITY ME Card T1E1 Single, issue command for the T1E1 software port
number as required.
• In ETERNITY ME use * command only if debug for all the T1E1 port is to be enabled/disabled.
Use following command to enabled T1E1 Port Level Debug (ETERNITY-ME and ETERNITY-GE)
6192-1-T1E1-1-Code
6192-2-T1E1-T1E1-1-Code
6192-*-1-Code
Where,
Code is from 000 to 255.
Default is 000.
Use following command to enable T1E1 Port-Port Level Physical Layer Debug:
6192 -1-T1E1- 2- Code
6192 -2-T1E1-T1E1- 2- Code
6192 -*- 2- Code
Where,
Code is from 000 to 255.
Step 10
Use the following command to program T1 Register Signaling Variant for the T1E1 port:
6161-1-T1E1-T1 Register Signaling Variant
6161-2-T1E1-T1E1-T1 Register Signaling Variant
6161-*-T1 Register Signaling Variant
Where,
T1E1 is from 1 to 8.
1 T1 RBS Decadic
2 T1 RBS DTMF
3 T1 RBS MFC R2
4 T1 RBS MFC R1
T1 RBS Decadic: A-bit is used to transmit the DNIS. The A-bit is toggle as per the Pulse dial Ratio Set.
T1 RBS DTMF: DNIS is transmitted in the corresponding speech channel using the DTMF signals as per the ITU-T
Q.23.
T1 RBS MFC R2: Same as above but using MFCR2 as per ITU-T Q.400-Q.490.
T1 RBS MFC R1: Same as above but MFCR1 as per ITU-T Q.300-Q.390.
Step 11
This parameter signifies the pulse dial ratio at which the signaling is done on the T1 RBS Line with T1 Register
Signaling = T1 RBS Decadic.
Use the following command to program digital pulse dial ratio for the T1E1 port:
6163-1-T1E1-Code
6163-2-T1E1-T1E1-Code
6163-*-Code
Where,
T1E1 is from 1 to 8.
Code Meaning
1 10 PPS, 1:2
2 10 PPS, 2:3
3 10 PPS, 1:1
Relevant Topics:
1. “T1 Maintenance” 1600
2. “T1E1 Trunks” 1608
What's this?
Certain features of the ETERNITY like Operator, Class of Service, Toll Control, Outgoing Trunk Access, among
others, require stations and trunks to behave differently according to the time of the day, which is referred to as
Time Zone.
For example, incoming calls are to be routed to the security personnel extension, instead of the Operator when the
office is closed, or certain features in the Class of Service are to be allowed only during working hours, or access to
outgoing long distance calls are to be denied during non-working hours, or the station must play a different greeting
message to the callers during break hours and holidays.
Time Tables can be assigned to stations and trunks to define their behavior according to the time of the day, i.e.
Time Zone.
Time Zones
A day can be divided into three time zones: Working hours, Break hours and Non-working hours. The default Time
Zones defined for each day are:
Working, Break and Non-Working hours are set to 00:00 for Sunday.
You can define a different Time Zone for your organization. Further, you can program each day of a week with
different time zones. For example, you may define the Working hours from Monday to Friday as 09:30 to 18:30, and
for Saturday, from 09:30 to 15:00. If you have a 24x7 business, you may set Working Hours also for Sunday.
Time Tables
A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours, Non-Working hours, for
the entire week.
A Time Table is assigned to stations defining the Time Zones for the entire week, so that the system can execute
the Time Zone-dependent features and facilities according to the Time Table.
Time Table 8
Time Table 7
Time Table 6
Time Table 5
Time Table 4
Time Table 3
Time Table 2
Time Table 1
By default, the Time Table 1 is assigned to all stations and trunks in their Station Basic Feature Template and
Trunk Feature Template respectively. In Time Table 1, six days of the week - Monday to Saturday -have working
hours from 9:00-18:00, break hours from 13:00-14:00 hours and non-working hours from 18:00 to 09:00. Sunday is
a holiday, with all three Time Zones set to 00:00 hours.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the stations
and trunks.
ETERNITY offers the facility to switch the system manually into "Day/Night mode", at any point in time, by
issuing a command. When you set the system in Day/Night Mode, the system overrides the Time Tables
assigned to Trunks, Stations and Operator. According to the mode you selected, it applies Working Hours/
Non-Working Hours to run all the Time-Zone dependent features of the system.
How to configure
A Trunk port can be assigned Time Table in the “Trunk Feature Template” assigned to it. A Station port can be
assigned a time table in the “Station Basic Feature Template” assigned to it.
The default Time Table 1 is assigned to both stations and trunks of ETERNITY. Check if this time table matches the
working hours of the organization, and the Time Zone requirements of the individual stations and trunks.
The following station parameters can be programmed differently for different Time Zones:
• Class of Service.
• Toll Control.
The following trunk parameters can be programmed differently for different Time Zones:
• DID
• DISA
The following features can be programmed differently for different Time Zones:
• Security Alarms
You may retain the default Time Table 1 or customize it to suit your requirements. Or you may customize different
Time Tables and assign them to different stations and trunks.
You can customize a Time Table using Jeeves or by dialing commands from a Telephone.
• Select the desired Time Table number and define the Time Zones, i.e. working hours, break hours and
non-working hours.
• Now, assign the Time Table you program to the desired Station/Trunk.
• To assign Time Table to Trunks, go to Trunk Feature Template. Refer the topic “Customizing Trunk
Feature Template using a Telephone” for details a how to assign timetable to trunks.
Start Time is Time Zone start time in 24-hours, HH:MM format. Where Hours is 00 to 23 and Minutes is
00 to 59.
End time is Time Zone end time in 24 hours, HH:MM format. Where Minutes is 00 to 59.
By default, Time Zone 1 is 0900 to 1800 and Time Zone 2 is 1300 to 1400. The left over time
automatically is treated as Non-Working Hours.
To assign Time Table to Stations and Trunk ports using SE commands, refer the topics “Customizing
Station Basic Feature Template using a Telephone” and “Customizing Trunk Feature Template using
a Telephone”.
• Exit SE mode.
What’s this?
The current time zone—Working Hours, Break Hours and Non-working Hours—is displayed on the LCD of the DKP
and the Extended IP Phone.
During Non-working hours the letter ‘N’ is displayed on the LCD of the DKP and the Extended IP Phone in the idle
state, and during Break hours, the letter ‘B’ is displayed.
During working hours, in the idle state, the phone display will look like this:
M on 20 D EC 16:58
3003 R eception 2
During Non-working hours, in the idle state, the phone display will look like this:
M on 20 D EC 16:58 N
3003 R eception 2
During Non-working hours, in the idle state, if the extension user has set User Absent and activated Keypad Lock
on the phone, the phone display will look like this:
What's this?
Toll Control (or Toll Restriction) is an expense control feature of ETERNITY. It enables you to program the system
so that each extension has a designated calling permission referred to as 'Call Privilege'.
Each type Call Privilege allows the extension to call certain areas and restricts it from calling others. The extension
can also be restricted from the dialing of specific telephone numbers.
• No Calls: Dialing of all external numbers is restricted. Only internal (extension-to-extension) calls are
allowed.
Only the numbers programmed in Global Directory Part I will be allowed to be dialed out, if the directory is
allowed in the Class of Service of the extension.
• Local Calls: Dialing of outgoing calls to Local area numbers, in addition to internal calls, is allowed. It is
possible to restrict calls to certain local numbers. To apply this Call Privilege, you must configure the 'Local
Numbers' list.
• Regional Calls: Dialing of outgoing calls to regional numbers is allowed, in addition to internal and local
calls. It is possible to restrict calls to certain regions. To apply this Call Privilege type, you must configure
the 'Regional Numbers' list.
• National Calls: Dialing of domestic, long-distance numbers within the country is allowed, in addition to
internal and regional calls. You can also restrict calls to certain parts of the country. To apply this Call
Privilege type, you must configure the 'National Numbers' list.
• International Calls: Dialing of international numbers is allowed, in addition to local area, long distance
and internal numbers. You can also restrict calls to certain countries. To apply this Call Privilege type, you
must configure the 'International Numbers' list.
• Limited Calls: Dialing of only specific Telephone numbers (regional, national or international) is allowed.
By applying this Call Privilege type, you can allow and restrict dialing of telephone numbers starting with a
particular digit, or a particular area code, or certain telephone numbers only. To apply this Call Privilege
type, you must program an 'Allowed List' with the numbers that are to be allowed and a 'Denied List' with
numbers that are to be restricted.
Toll Control forms the basis of the features “Dynamic Lock” and “Call Budget”.
Using “Dynamic Lock”, extension users can change the Toll Control (Call Privilege) of their extensions on their own.
The Operator or System Administrator can also change the Toll Control of the extension using Dynamic Lock. To
support this feature, ETERNITY offers fours levels of Toll Control, from 0 to 3.
When the “Call Budget” feature is used on extensions, it becomes necessary to define the calling permission for
extensions that have consumed their allotted budget. To support this feature, ETERNITY offers Toll Control-Call
Budget Consumed.
• Toll Control - Level 0 is Time Zone based, wherein you must define the Call Privilege Type for each Time
Zone, i.e. Working Hours, Break Hours and Non-Working Hours. For instance, you may define
'International Calls' as Call Privilege for Working Hours, 'Local Calls' as Call Privilege for Break Hours and
'No Calls' as Call Privilege for 'Non-Working' Hours.
By default, Call Privilege 'International Calls' is selected for all three Time Zones.
• Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
• Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
• Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
• Toll Control - Call Budget Consumed is applied only if the “Call Budget” feature is enabled on the
extension.
ETERNITY offers you the flexibility to redefine the Call Privilege for each of the above Toll Control Levels according
to user requirements.
How it works
• When a call is made, the ETERNITY checks the Toll Control Level assigned to the extension making the
call.
• The system checks the 'Call Privilege' programmed in the Toll Control Level of the extension.
• For each call privilege type detected, the system will check the following to determine if call is to be
allowed or denied, as summarized in the table below:
Limited Calls Allowed and Denied Number Lists assigned to the extension in its Station
Basic Feature Template
• The Local, Regional, National and International Number Lists consist of Allowed Numbers and Denied
Numbers.
• matches with Allowed Number list and the Denied Number list.
• matches with Allowed Number list, but not with the Denied Number list.
• matches with neither the Allowed List nor the Denied List.
• The call is restricted, if the dialed number matches with the Denied Number list, but not with the Allowed
Number list.
How to configure
For Toll Control to work, you must first program the lists of Local Numbers, Regional Numbers, National Numbers
and International Numbers. With these number lists ready, you may program the Toll Control Levels.
Before you program, make a two-column tables each for Local, Regional, National and International numbers on
paper or using a computer. On one column of each list, write down the numbers you want to permit. On the other
column write down the numbers you want to restrict.
999
999
999
• Enter the local area numbers that are permitted to be dialed in the 'Allowed List' and the numbers that are
to be restricted in the 'Denied List'. You may enter as many as 999 numbers in each list.
• Enter the regional area numbers that are permitted to be dialed in the 'Allowed List' and the numbers that
are to be restricted in the 'Denied List'.
• Repeat the entries you made in the 'Local Number' list also in the 'Regional Numbers' list.
• Enter the long distance numbers within the country that are to be permitted in the 'Allowed List' and the
numbers that are to be restricted in the 'Denied List'.
• Repeat the entries you made in the 'Local Numbers' and the "Regional Numbers' lists in this list.
• Enter the overseas numbers that are to be permitted in the 'Allowed List' and the numbers that are to be
restricted in the 'Denied List'.
• Repeat the entries you made in the lists of 'Local Numbers', 'Regional Numbers' and 'National Numbers' in
this list.
• By default Number List 01 is assigned to both Allowed and Denied List for Limited Calls.
• Select a different Number List for Allowed List, for example 02. Enter the specific numbers or digits that are
to be allowed to be dialed in the list.
• Select another Number List for Denied List, for example 03. Enter the specific numbers or digits that are to
be restricted from being dialed in this list.
It is not mandatory to assign the same Limited-Calls Allowed-List and Denied-List for all Time Zones of Toll
Control Level 0 or to other Toll Control Levels. You can prepare different Allowed and Denied Lists for
each Toll Control Level.
• By default all stations of ETERNITY are assigned Template 01. You may customize this template or select
another template.
• Select the desired Call Privilege Type for each Time Zone - Working Hours, Break Hours, Non-Working
hours.
• Repeat the same steps to select the Call Privilege type for other Toll Control Levels 1, 2 and 3.
Local Numbers
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
Regional Numbers
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
National Numbers
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
International Numbers
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
Number is the number string you want to store at this location index on the list. The number can be
maximum 16 digits and should be terminated with #* if it has fewer than 16 digits. Refer the following
table for codes for dialing special digits: 0-9, #, *, A, B, C, D, Flash (F), Pause (P), +, Dot (.).
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls
6 for Limited Calls
0 for No Calls
1 for Local Calls
2 for Regional Calls
3 for National Calls
4 for International Calls
0 for No Calls
1 for Local Calls
2 for Regional Calls
3 for National Calls
4 for International Calls
0 for No Calls
1 for Local Calls
2 for Regional Calls
3 for National Calls
4 for International Calls
0 for No Calls
1 for Local Calls
2 for Regional Calls
3 for National Calls
4 for International Calls
• Exit SE mode.
Also refer the topics “Configuring Extensions”, “Station Basic Feature Template”, “Number Lists”.
The topic ‘Trunk Access Group’ is renamed as ‘OG Trunk Bundle Group (OGTBG)’ in Firmware Version V6 and
later of ETERNITY.
What’s this?
• The calls landing on a trunk are answered automatically.
• By enabling Auto Answer on trunks, you can ensure the caller will not get a busy tone, which may force
him to disconnect his call.
• By enabling Auto Answer, the caller will be greeted with pleasant greetings and voice messages before the
call is actually handled. The caller will get music when station becomes free.
• By enabling Auto Answer, the caller will be offered ‘Busy Message’ called ‘Trunk Auto Answer Busy Bye
Message’, if the incoming call is answered using Trunk Auto Answer feature and the call is not answered
by routing group member within specific time of DID inactivity Timer. Refer flow chart.
How it works
• ETERNITY supports 3 types of Trunk Auto Answer.
• Disabled.
• All (All calls landing on the trunk are answered automatically).
• Busy.
• The Trunk Auto Answer type can be different for different Time zones viz. WH, BH, NH.
• Greetings can be assigned to the trunks to greet the caller based on the Time Zone.
• After the greetings the trunk can be programmed to play one of the following to the Caller.
• ETERNITY offers 4 Voice Messages that can be played back after the greetings.
• ETERNITY offers 4 voice messages for Trunk Auto Answer Busy Bye Message.
How to configure
Please refer the topic “Trunk Feature Template” for details on programming Trunk Auto Answer.
This feature is same as the one offered by many IVR systems, like Railway Enquiry, Banking Industry, etc.
This notifies the caller that someone would attend him shortly. This feature finds large application in call
centers.
Auto
No
Answer
Enable?
Yes
Process the call as
Answer the call per the programming
Call No
Answered?
End
Relevant Topics:
“Trunk Feature Template”
“Voice Message Applications”
“Auto Answer”
Trunk Auto Answer is useful when you want callers to remain connected until one of the landing destinations
selected for incoming trunk calls becomes free to attend to the caller.
Trunk Auto Answer is useful in call centres, railway enquiry, banks, where callers need to be notified that they
would be attended shortly, so that they do not disconnect the call.
• For all Calls: the system answers all incoming calls landing on the trunk line.
• When Busy: the system answers incoming calls on the trunk, only if the landing destinations are busy.
How it works
Trunk Auto Answer on a trunk works only when Operator or Extension/s is selected as the landing destination for
incoming calls for the Day and Night.
So, you can enable Trunk Auto Answer on a CO, Mobile or SIP trunk, only if you have selected Operator or
Extensions as the destination for incoming calls on that trunk.
ETERNITY NE handles incoming calls on the trunk according to the type of Trunk Auto Answer selected for the
trunk: For all Calls or When Busy
When Trunk Auto Answer–For all Calls is enabled on a CO, Mobile or SIP Trunk, for each incoming call on the
trunk,
• The System answers the with a Greeting message, known as the Trunk Auto Answer Greeting, and rings
the landing destination—Operator or Extensions—selected for the time of the day.
The system starts the DID Inactivity Timer (default: 60 seconds). The Trunk Auto Greeting message is
played once. You may assign a Trunk Auto Answer Greeting of your preference.
• If the landing destination does not answer before the Trunk Auto Answer Greeting message ends, the
system plays Trunk Auto Answer Ring Back Tone message to the caller.
The Ring Back Tone message is played repeatedly for the duration of the DID Inactivity Timer.
However, if no Trunk Auto Answer Ring Back Tone message is assigned, the system will plays Ring Back
Tone to the caller for the duration of this timer.
• If any of the landing destinations answers the call before the expiry of the DID Inactivity Timer, the system
stops the DID Inactivity Timer and the Ring Back Tone message, and connects the caller to the extension
that answered the call.
If no Trunk Auto Answer Busy Bye message is assigned, the system plays the Busy Tone for the duration
of the Busy Tone Timer and releases the trunk port.
When Trunk Auto Answer–When Busy is enabled on a CO, Mobile or SIP Trunk, for each incoming call on the
trunk,
• The System answers the with the Trunk Auto Answer Greeting message and loads the DID Inactivity
Timer.
• The System waits for any of the landing destinations (Operator or Extensions) selected for the time of the
day to be free.
• If no landing destination is free at the end of the Trunk Auto Answer Greeting message, the system plays
Ring Back Tone or Trunk Auto Answer Ring Back Tone message, if assigned, to the caller for the duration
of the DID Inactivity Timer.
• If any of the landing destinations is free before the expiry of the DID Inactivity Timer, the system places the
call on that destination.
• If none of the landing destinations is free at the end of the DID Inactivity Timer, the system plays the Trunk
Auto Answer Busy Bye message, if assigned, and releases the trunk port.
If the Busy Bye message is not assigned, the system will play the Busy Tone to the caller for the duration
of the Busy Tone Timer.
How to configure
For this feature to work, you must do the following:
1. Enable Trunk Auto Answer on the CO Trunk, Mobile, SIP trunk, as required.
• Under Basic Settings, open the links to the desired trunk port type.
• Open the Route Incoming Calls link on the page of the selected trunk number.
• If Operator or Extensions is selected as the destination to Route calls during the Day and to Route
calls During the Night, configure Trunk Auto Answer.
It is not mandatory to configure Trunk Auto Answer for both time zones. You may configure for any time
zone, according to your requirement.
• Select the Trunk Auto Answer Greeting message, the Trunk Auto Answer Ring Back Tone
Message, and the Trunk Auto Answer Busy Bye Message for the Day and Night.
You may select different Greeting, Ring Back Tone and Busy Bye Message for each time zone.
2. Configure the Trunk Auto Answer related Timers, if required. The following Timers are of relevance to the
Trunk Auto Answer Feature:
• The DID Inactivity Timer (default: 60 seconds)
• The Ring Back Tone Timer (default: 45 seconds)
• The Busy Tone Timer (default: 7 seconds)
• You may change the duration of these timers from the“System Timers and Counts” page under
Advanced Settings.
The Ring Back Timer and the Busy Tone Timer are also applicable for the Ring Back Tone and the Busy
Tone played for internal calls.
3. Record and assign Voice Modules for the following Voice Messages related to this feature:
• Trunk Auto Answer Greeting Message.
• Trunk Auto Answer Ring Back Tone Message
• Trunk Auto Answer Busy Bye Message
For each of these messages, you can record four different messages.
If you do not want to use messages, you may select the options: Music-on-Hold or Do not Play any
message.
What’s this?
A Trunk Landing Group is a group of extensions on which incoming calls on a particular trunk are landed.
Trunk Landing Groups are formed for efficient call management. Generally, incoming calls on a trunk are landed on
the Operator extensions. However, when several trunks are interfaced with the system, it becomes difficult for the
operator to answer all calls efficiently. Trunk Landing Groups relieve the Operator to a great extent, as the incoming
calls get distributed among several extensions.
How it works
• A Trunk Landing Group (TLG) is a “Routing Group”.
• You can configure as many as 95 TLGs. Each group is numbered from 01 to 95.
• A maximum of 32 stations—SLT, DKP, SIP, ISDN, or DOP—can included in each Trunk Landing Group.
• For each group that you create, you can do the following:
• set the Sequence in which the stations in the group should ring, by selecting the member stations in a
sequence from 1 to 32.
• set the Time for which each station in the group should ring, by setting the Ring Timer (default: 15
seconds).
• set each station to ring continuously till the call matures by enabling Continuous Ring (default:
disabled).
When Continuous Ring is enabled, once a station receives a ring, it rings continuously till the call
matures. The station continues to ring even as other stations in the group are hunted.
If the call is not answered even after the last station in the group has been hunted, the system will loop
back and start hunting from the first station, all over again.
• have a number of stations in the group ring simultaneously by enabling Continuous Ring on these
stations and setting the Ring Timer for these stations to ‘00’ seconds.
• set equal distribution of incoming calls on all stations in the group, by enabling Rotation for the entire
group (default: disabled).
When Rotation is enabled on a TLG, for each new call on a trunk, the system will land the call on the
extension next to the one that received the last call.
When Rotation is disabled in a TLG, for each new call on a trunk, the system will land the call on the
first free station of the TLG.
• To each Trunk, you must assign a TLG for the Time Zones, working hours, break hours and non-working
hours. You may assign the same TLG for all three Time Zones, or a different TLG for each Time Zone.
• Configure each TLG as a Routing Group. See “Routing Group” for instructions.
• Assign the TLGs you formed for each trunk for the three Time Zones in its Trunk Feature Template. for
instructions, see “Trunk Feature Template”.
Example:
Two TWT Lines (configured on software ports 001 and 002) are interfaced with ETERNITY.
• Incoming calls on TWT 1 during working hours should land on SLT stations 2001, 2003, 2005 (configured
on software ports 008, 010, 012 respectively).
Incoming calls on TWT 1during break hours and non-working hours should land on SLT stations 2002,
2004 (configured on software ports 009, 011 respectively).
• Incoming calls on TWT 2 should land always on DKP stations 3001, 3002, 3003, 3009, 3010 (configured
on software ports 013, 014, 015, 016, 017).
• Incoming call on TWT 1 should ring for 10 seconds on each station in the TLG.
• Incoming call on TWT 2 should ring for 20 seconds on each station in the TLG.
• The stations of the TLGs of TWT 1 and TWT2 should ring for the set time only.
In this example, for TWT 1, you would need to form 2 TLGs; one TLG for working hours and one for break hours
and non-working hours. So, form two Routing Groups. For example, Routing Group 10 for working hours and
Routing Group 11 for break hours and non-working hours.
In Routing Group 10, select the member SLTs in this sequence: 2001, 2003 and 2005. Set the Ring Timer for each
member SLT to 10 seconds. Disable Continuous Ring for the SLT, as each station in the group must ring for the set
time.
In Routing Group 11, select the member SLT in this sequence: 2002 and 2004, and set the Ring Timer for each SLT
to 10 seconds. Disable Continuous Ring for the SLT, as each station in the group is required to ring for the set time
only.
For TWT 2 you would need to form a common TLG for working hours, break hours and non-working hours. So,
configure as single routing group, for example, Routing Group 13 for TWT 2. In Routing Group 13, select the
member stations in this sequence: 3001, 3002, 3003, 3009, 3010. Set the Ring Timer for each DKP member station
to 20 seconds. Disable Continuous ring for each DKP in the group. Enable Rotation for Routing Group 13.
In the Feature Template of TWT 1, assign TLG. For working hours, assign Routing Group 10, for break and non-
working hours, assign Routing Group 11.
In the feature Template of TWT 2, assign Routing group 13 as TLG for all three time zones.
What’s this?
This feature enables any extension user to reserve a trunk for exclusive use, for a specific time period.
Trunk Reservation can be requested from an SLT extension, a DKP extension and from an Extended IP Phone
extension.
How it works
Let us understand this feature with the help of an example:
In an organization there are four TWT trunk lines, TWT 1, 2, 3 and 4, but all of these have full traffic throughout the
day.
Extension user A is a sales executive. To complete the sales target, A needs to make long-distance calls to
customers. Since there this full traffic on all the four trunks throughout the day, and these trunks are constantly
busy, A would need a dedicated trunk line to save time and complete the target.
So, A can reserve one of the four trunk lines for the desired duration. To do this,
• Extension A dials the feature access code for Trunk Reservation for the busy trunk.
• A answers the call and gets connected to the trunk, and gets dial tone.
• The trunk remains reserved for the duration of the Trunk Reservation Timer. This timer is configurable, and
by default is set to 10 minutes. A can have this Timer configured to the desired duration.
• All other extension users who try to access this trunk get error tone, even if this trunk is free.
• If A is finished with the calls before the expiry of the Trunk Reservation Timer, A has two options:
a) release the trunk manually, by cancelling Trunk Reservation.
or
b) wait for the expiry of Trunk Reservation Timer.
• Only when the trunk is released (by A or at the end of the Timer) will other users be able to access it.
You may increase or decrease the duration of the Trunk Reservation Timer. See “System Timers and Counts” for
instructions.
How to use
For EON & Extended IP Phone Users
OR
To release a reserved Trunk, wait for the Timer to expire, or cancel Trunk Reservation, manually.
OR
• Dial 102
• Lift handset.
• Dial 6 on Busy Tone.
• Replace handset.
What's this?
Extension users may sometimes want to leave their desks, and expecting to return soon, they may not have
forwarded their calls or set Do Not Disturb on their extensions. In such cases, incoming calls will continue to land on
the extension and go unanswered. The callers have no way of knowing that the extension user is not present at the
extension and may try the extension number repeatedly.
With the User Absent/Present feature of ETERNITY, extension users, including the Operator, can set 'User Absent'
when they leave their desks. By doing so, they can block all incoming external as well as internal calls from landing
on their extension. When they return to their desks, they can set 'User Present' and receive incoming calls again.
While Do Not Disturb blocks only internal calls, User Absent can be set when you want to block all
incoming calls (external as well as internal). Thus User Absent can be used as an extension of DND to also
block external calls.
There are more options for indicating availability to other extensions. Refer the topic “Presence” to know
more.
How it works
When an extension user of EON sets 'User Absent', the letter 'A' appears on the phone's display:
EON42 EON48P
617 Himanshu Fri 29 JAN 16:58 A
Fri 29 JAN 16:58 A 617 Himanshu
The letter 'A' disappears when the extension user sets 'User Present'.
When an extension user of EON calls the extension which has set 'User Absent', the text message 'User Absent'
will appear on the caller's phone display.
When an SLT extension user calls the extension which has set 'User Absent', callers who dial this extension will get
an error tone.
External callers who call the extension, on which 'User Absent' is set, will get an error tone only.
• Outgoing calls can be made from the extension which has set 'User Absent'. Only incoming calls are
restricted.
• User Password is required for this feature. The default User Password, 1111, will not work. Change the
User Password first.
How to use
The System Administrator can also set an extension as Absent/Present using Jeeves. For instructions, read
“Additional Security to Extension Users” under the topic System Security.
What’s this?
The User Password is a 4-digit code for extension users to protect their extension phones from unauthorized use.
The default User Password is 1111. It can be changed by the extension users from their phones to any desired
value, not exceeding 4 digits.
In case the extension user forgets the password, it can be cleared and restored to the default value 1111 by the
System Engineer (SE) or the System Administrator (SA). Refer the topic “System Security” for instructions.
The User Password is also required to access and use certain features of ETERNITY, which are listed below.
• Call Follow Me
• Dynamic Lock
• Direct Inward System Access (DISA)
• Walk-In Class of Service
• User Absent/Present
• Hot Desk
• Phone Settings of the DKP
• Mailbox of Voice mail
The extension user must change the default password for all the above listed features except: Phone Settings,
Mailbox of Voice mail. Both these features allow the extension user to use the default User Password, whereas in
the case of others, the system will not allow feature access without changing the User Password.
In the case of Hot Desking, the default password will work only for one extension involved.
The User Password for an extension can be changed only from that extension phone only.
Since the Mailbox can be accessed using the default User Password, extension users who are assigned a
mailbox are recommended to change their User Password to prevent unauthorized access to their mailbox.
How to use
What’s this?
• Video call can be set up if the service provider supports it.
• The Video Conferencing equipment (VC) is connected at the NT Port of the PBX.
• OG Call
• IC Call
• Conferencing
Relevant Topics:
1. “ISDN-BRI” 1263
2. “ISDN-PRI” 1285
What’s this?
The Virtual Station feature of ETERNITY enables multiple users to share one telephone instrument as their
extension, yet be considered as individual extensions by the system, with distinct extension properties and class of
service.
Such shared extensions are called Virtual Stations, as their users do not have individual phones for their use.
Virtual stations are useful in laboratories, common rooms, dormitories, shop floors, and wherever it is not feasible
to provide dedicated telephone instruments to individual extension users. Virtual extensions allow you make
optimum use of the existing phones without investing in new ones.
How it works
The shared telephone instrument is called the Master Station. A Master Station can be an SLT, a DKP or the Matrix
Extended IP Phone.
• Virtual Stations are assigned to the Master Extension. A Master Extension can have multiple Virtual
Stations, but a Virtual Station can have only one Master Station.
• The Virtual Station functions as any SLT, DKP or SIP extension of ETERNITY. It can be assigned all
features and facilities, like Class of Service, Toll Control, Call Forward, just like any other physical
extension of ETERNITY.
• All incoming, outgoing, internal and external calls of the Virtual Stations are recorded in the Station
Message Detail Records.
• To make outgoing calls, the Virtual Station user must use the feature “Walk-In Class of Service”.
• The Virtual Station user is logged out of the Master Station according to the Walk Out mode assigned to it:
Walk out single call or Walk out multiple calls.
How to configure
To use this feature, you must assign the Master Station and the Master Port for the extensions you want to
configure as Virtual Extension. For instructions, see “Station Advanced Feature Template”.
How to use
For making outgoing calls users of Virtual Extensions must use “Walk-In Class of Service”.
What’s this?
The Voice Help feature of ETERNITY allows you to record and play short (16 second duration) voice messages to
provide quick help to extension users. Voice Help messages may contain important instructions, or frequently
accessed feature codes, or important phone numbers, etc.
For example, the Access Codes of the frequently used features can be recorded in the Voice Help message.
Extension users may simply dial the Voice Help feature code and listen to the voice message.
How to configure
To be able to use Voice Help, you must first record a voice module with the contents you wish to provide as help.
Record the voice module considering the maximum duration of the voice module, so that the message is not
truncated.
The voice module must be assigned to the Voice Help application. Refer topic “Voice Message Applications” for
instructions on recording the voice module and assigning it to Voice Help.
How to use
What’s this?
Any Voice Mail System or Auto-Attendant can be integrated with the ETERNITY. However this integration works
well if the exchange of information between the two is perfect. Generally, this integration works on two protocols:
Tone Sensing and Gateway.
Tone Sensing relies on sensing of various Call Progress tones. Gateway uses DTMF digits to signify various Call
Progress events. The ETERNITY supports both Tone Sensing as well as the Gateway mode.
User can transfer the call on the mail box of the station by dialing the access code for 'Blind transfer to VMS'. Refer
chapter “Call Transfer” for more details.
ETERNITY supports:
• External VMS - Matrix CadencePro or any third-party external VMS system.
• ETERNITY ME/GE/PE Card VMS16
See the System Manual for 'ETERNITY VMS Card' for more information.
See the ETERNITY Hospitality System Manual for more information about Hotel/Motel applications of this
feature.
External VMS
The figure given below shows connection of a VMS with the ETERNITY where only two VMS ports are interfaced.
How it works
As shown in the figure, VMS is connected on two SLT stations. If all the calls are to be routed through VMS,
ETERNITY should be programmed such that all the incoming calls land on the VMS. Alternatively, the PBX should
be programmed such that incoming calls land on other extensions.
As stated earlier, integration of the ETERNITY with VMS or an Auto-Attendant works on two protocols: Tone
Sensing and Gateway.
Tone Sensing relies on sensing of various Call Progress tones. For example, when an incoming call lands on VMS
through the ETERNITY, it answers the call and prompts the caller to dial a station.
Subsequently, it senses the dialed station number, keeps the caller on hold and dials the station number where the
call is to be diverted. Then the VMS waits for the ring back tone. It first senses the ring back tone and then waits for
it to stop.
As soon as the ring back tone stops the VMS transfers the call to the dialed station. All this process involves
sensing of various tones like ring back tone, hold tone, etc. and hence is commonly called tone sensing. However,
this method is not accurate.
Gateway uses DTMF digits to signify various Call Progress events. With Gateway mode, in the above example
when the VMS receives an incoming call, it answers the call. Then it prompts the caller to dial a station. It senses
the dialed station, keeps the caller on hold and dials the station where the call is to be diverted. On successful
receipt of the dialed station, the ETERNITY sends a defined code (signifying ring back tone) to the VMS.
Depending upon the transfer type set for the VMS, it either transfers the call or waits for the called party to respond.
In the latter case, when the called party answers the call, the ETERNITY again sends a defined code (signifying
successful speech connection). The VMS detects this code and transfers the call to the dialed station. Since codes
(defined string of characters) are used for the transfer of information, this method has much higher reliability.
• ETERNITY will automatically configure required number of SLT port as VMS ports.
• The port name of all these VMS SLT Ports is assigned as 'Voice Mail'.
• The ETERNITY ME/GE/PE Card VMS supports maximum 512 mailboxes and each mailbox can store total
of 254 messages.
• The System shall automatically assign VMS group when ETERNITY ME/GE/PE Card VMS is installed.
The default VMS Group assigned is Routing Group number '96'. Refer chapter “Routing Group” for more
details.
• SA can place the Voice guided remote alarm call to the voice mail. Refer chapter “Alarms” and “DSS Keys
Programming” for details of SA command.
From SA mode:
Code Meaning
How to program
Default IP Address of the Ethernet port of the ETERNITY ME/GE/PE Card VMS 16, is 192.168.001.131.
Step 1
Use following command to assign Destination Type for VMS group:
4601-Destination Type
Where,
1 Routing Group
2 Number
Step 2
Use following command to assign Destination for VMS group:
4602-Destination-#*
Where,
Destination is 10 digit maximum with digits 0-9, * and #.
By default, Destination is '96'.
Please refer the topic “Routing Group” for details on programming. Take care to see all the stations programmed in
the Routing Group assigned as VMS Group are SLTs. Assigning 00 as routing group means no routing group is
programmed as VMS Group.
Access code is the code dialed from the dial phase to call the VMS group. The VMS group access code must be
unique and should not match with either SLT station number or DKP station number or any of the features available
from the dial phase.
Use the following command to program the access code for a VMS Group:
3114-1-VMS Group Index-Access Code-#*
3114-2-VMS Group Index-VMS Group Index-Access Code-#*
3114-*-Access Code-#*
Where,
VMS Group Index is 1.
Access Code is maximum of 6 digits. If it is less than 6 digits, terminate it with #*
By default, the Access Code of the VMS Group is 3931
Use following command to default the access code for a VMS Group:
3164-1-VMS Group Index
3164-2-VMS Group Index-VMS Group Index
3164-*
Where,
VMS Group Index is 1.
Use following command to clear the access code for a VMS Group:
3114-1-VMS Group Index-#*
3114-2-VMS Group Index-#*
3114-*-#*
Where,
VMS Group Index is 1.
Step 4
Use following command to know 'IP Address: Web Server Port' of VMS Card:
2131-Slot Number
Where,
Slot Number is from 01 to 16 (In which VMS Card is inserted).
Relevant Topics:
“Routing Group”
“SLT Hardware Template”
“Message Wait”
“QSIG”
“Call Transfer”
What’s this?
ETERNITY allows you to record different voice messages which can be played to callers/extension users according
to the situation. For example, if you have activated “Direct Inward Dialing (DID)” on some of the trunk lines, you can
use an appropriately recorded Voice Message to guide callers to reach the desired party/destination extension.
Similarly, an extension user who wants to set an alarm can record a personal voice message on his/her own, which
will be played to him/her when the alarm request is served.
How it works
The voice messages are recorded in 'Voice Modules' and the voice modules are assigned to the features/
applications with which they are to be used.
The ETERNITY supports 16 Voice Modules of a maximum duration of 16 seconds each. You can record up to 16
short messages of a maximum 16 seconds each or less.
• Once-Only: the message is played only once from its start to its end.
• Continuous: the message is played repeatedly from the start to the end.
When the recorded voice modules are assigned to the features/applications, they are played to the callers/
extension users whenever the feature/application is activated.
As many as five different voice messages can be played simultaneously to a caller/extension user.
Voice messages can be used for different applications or situations as described in the following.
For example:
• During Greeting message lunch hours: "Welcome to Cotton Software. This is lunchtime. Please call
after 2.00 pm".
• DID Greeting Message for non-working hours: "Welcome to Cotton Software. We are closed for the
day. Please call later".
For example,
• DID Wrong Dial Message: "Sorry you have dialed an invalid Station Number. Please dial a valid Station
Number".
• DID Ring Back Tone (RBT) Message: "You have successfully dialed the desired Station".
• DID Busy Message: "The station you have dialed is busy. Please hold the line or try any other Station
Number".
• DID No-Reply Message: "The Station you have dialed is not responding. Please try another Station".
• DID No-Dial Message: "Sorry you have not dialed any Station Number. Please wait while your call is
transferred to the Operator".
• DID Call Transfer Message: "Transferring the call to the Operator". This message is played to the caller,
when he does not dial any number and the call is transferred to the Operator.
For example, "Good Morning. This is a wake up call. You may please call room service for any assistance.
Thank you. Have a nice day".
You are recommended to record the message "Please press 0 to acknowledge the Alarm and Reminder"
as a voice module for the Alarm/Reminder message, so that extension users can acknowledge Snooze
calls. Refer the topics “Alarms” and “Reminder” to know more.
• Security Alarm: A pre-recorded voice message is played to the external number and to the extension
which answers the Security Alarm call. The default message played to the called parties is: "This is an
emergency call. Please dial '0' to acknowledge.
• Voice Help: You can record and play voice message to provide quick help to extension users. The Voice
Help message may contain important instructions, or frequently accessed feature codes, or important
phone numbers, etc.
Help message must be recorded considering the maximum duration of the Voice Module (16 seconds), so
that voice messages are not truncated.
• Music-on-Hold: Callers who are put on hold are usually played music from an internal/external source as
they wait. You can play a voice message instead of music to the callers. The message may contain any
promotional information about your company or services provided by your organization, etc.
For example, "Welcome to Progressive Bearings. We are glad to announce that we are now an ISO 9001
company."
• Message Waiting: Whenever there is a new message in the mailbox of the extension user and if the VMS
informs the ETERNITY about the new message, the ETERNITY changes the dial tone of the extension to
a stuttered dial tone. The ETERNITY also offers the facility to playback a message instead of the stuttered
dial tone to indicate the waiting message. An appropriate voice message can be played back to the
extension user when he lifts the handset.
For example, "You have a new message in your Mailbox. Please access your mailbox".
For example, "Please dial the number immediately after the beep".
When you record a message for the Dial Tone, make sure that the length of the message is shorter than
the duration of the Dial Tone Timer.
When you record the Ring Back Tone message, take care that the length of the message is shorter than
the duration of the Ring Back Tone Timer.
When you record the Busy Tone message, take care that the length of the message is shorter than the
duration of the Busy Tone Timer.
Voice Message Notification for DND is particularly useful when the phone on which DND is set is an SLT and
callers are also using SLT.
Similarly, a voice message can be played to the extension user who attempts to invoke a feature that is not allowed
to his/her extension phone in its Class of Service (COS).
These time based greetings are played to callers before the DID Greetings Message on a DID enabled trunk. On
an Auto-Answer enabled trunk these messages are played before the trunk AutoAnswer greetings.
How to configure
To be able to play Voice Messages, you must first record them in voice modules. Once you have recorded the
voice messages, you must assign the voice module to the appropriate Voice Message Application.
Pre-recorded voice messages are provided in WAV format on the documentation CD shipped with the ETERNITY.
You may either use them when recording the voice modules or you may record messages of your choice.
• An External Music Source (PC, Music System) connected to the “Analog Input Port (AIP)” of the
ETERNITY. For instructions, refer the topic Connecting External Music Source under “Installing ETERNITY
ME”, “Installing ETERNITY GE”, “Installing ETERNITY PE” as relevant to your model of ETERNITY.
If you want to use the pre-recorded voice messages provided with the system, first copy the WAV files on
to the PC to which ETERNITY is connected. The contents of the audio files are indicated by their file
names. Select the audio files (containing the messages) you wish to use, and copy them on to the PC.
Play the files when you record the messages. Recording instructions are provided below.
• 2503-Voice Module
Where,
Voice Module is from 01 to 16.
In this case, dial the number of the Voice Module you just recorded.
If the audibility of the recorded message is not satisfactory, you may repeat this procedure again.
• Recording Source: Select the source of recording - Telephone Instrument or Music System. By
default, Telephone instrument is selected.
• Recording Format: Select the format of the voice message recorded, A-Law, Mu-Law or Linear, as
appropriate. By default Mu-Law is selected.
• By default, the following Voice Messages Applications have been assigned to Voice Modules 01 to 13
(see table below). If the default Message recorded in the module suits your purpose, simply assign the
Voice Module number to the relevant Voice Application. The system will automatically set the duration
of the Voice Message Application.
• For example, if you want to use Morning, Afternoon and Evening Greetings. You may simply assign the
default Voice Modules 02, 03, and 04.
• Refer the following table for Voice Message Applications assigned by default to the Voice Modules:
Voice Module
Voice Message Application Voice Message
Number
01 Music-On-Hold
06 DID Welcome Greeting for Night time Welcome! I am sorry, we are closed.
(Non-working and Break hours)
08 DID - No Dial message Sorry! You have not dialed any number.
11 DID - Destination Ringing message The number you have dialed is ringing.
(Ring Back Tone)
12 DID - Destination No Reply message The person you dialed is not responding.
13 DID Call Transfer to Operator message Please hold, transferring your call to the Operator.
• If you have already recorded Voice Module 08, ETERNITY will automatically detect and display the
duration of the Voice Module you recorded. So you need not define the duration of the Voice Module.
• You may define the duration of the Voice Module, only if you want the recorded voice message to be
played for a specific duration. For example, the message you recorded in the voice module is 15
seconds long, but you want to play only the message contents of the first 8 seconds, you can define the
duration of the message as 8 seconds.
• Voice Module 01 is reserved for Music-on-Hold by default. You are advised not to assign this module to
any other Voice Message Application.
32 Alarm Continuous
To use the default Voice Modules, refer the table “Default Voice Message Applications assigned to Voice
Modules”, and assign the desired Voice Module to a Voice Message Application.
• Exit SE mode.
Code is
0 for Cancel
1 for Set
What’s this?
Every extension of ETERNITY is allowed a distinct Class of Service and Toll Control defining its access to features
and its calling permission.
Extension users may be required to make calls from another extension, which does not have the same Class of
Service and Toll Control as their own extension.
With Walk-In Class of Service, all extension users of ETERNITY can make calls from any other extension of the
system as per the Class of Service and Toll Control of their own extension.
This feature is particularly useful to extension users who frequently move away from their desk. It allows them the
same level of feature access and Toll Control on the other extensions from which they make calls.
Extension users can 'Walk-In' from any extension port: DKP, SLT, ISDN Terminal, Magneto, E&M (with Station as
Orientation Type), and SIP.
• One call: The extension user is automatically logged out after s/he has made a call from the extension into
which s/he has walked in.
• Multiple Calls: The extension user can make as calls as desired, and remains 'walked-in' until s/he
manually 'Walks-Out' or until a second person walks into the same extension port.
Extension users can be allowed either of the above types of Walk-In by programming the 'Walk-Out Mode" in their
“Station Advanced Feature Template”.
Walk-In Class of Service is a password-protected facility. The User Password is required to operate this feature.
The default User Password 1111 must be changed for using this feature.
How it works
• Extension user A has a DKP with the number 3001. He has long distance calling facility on his extension.
• Extension user B has an SLT with the number 2001. She can make only local calls from her extension.
• Extension user A is at B's desk and needs to make a long distance call.
• Extension user A can 'Walk-In' into B's extension number 2001 by dialing
• the feature code for 'Walk-In Class of Service'.
• the User Password (default password 1111 will not be accepted).
• On successful Walk-In, the system applies the Class of Service and Toll Control of extension 3001 on
extension 2001.
• Extension user A can now dial the trunk access code and the external, long distance number.
• If Extension 2001 has 'Multiple Calls' selected as the 'Walk-Out Mode' for the extension, extension user A
must manually walk out by dialing the feature code for 'Walk-Out'.
• If Extension user A does not 'Walk-Out', the system will perform a walk out for A only when another
extension user walks into extension 2001.
• Calls made by Extension user A from extension 2001 using Walk-In will be calculated and charged to
extension 3001 only.
• Call record details of calls made by Extension user A from extension 2001, using Walk-In will be recorded
in the Station Message Detail Record of extension 3001 only.
At a time, only one extension user can walk-in into another extension.
How to configure
This feature is available to all extensions of ETERNITY. It does not require any other programming except selection
of the 'Walk-Out Mode' for the extension in its “Station Advanced Feature Template”.
By default, Station Advanced Feature Template 01 is assigned to all extensions of ETERNITY and the default
'Walk-Out Mode' selected in the template is 'One Call'. Retain this template if you want to keep this as the Walk-Out
mode for all extensions.
If you want to allow different walk-out modes to different extensions, then decide which of the extensions are to be
allowed 'One Call' and which are to be allowed 'Walk-In until Logout'.
For each extension, select the 'Walk-Out Mode' in its Station Advanced Feature Template. You can do this using
Jeeves or by dialing SE commands from an extension phone.
For detailed instructions refer the topics “Customizing Station Advanced Feature Template using Jeeves” and
“Customizing Station Advanced Feature Template using a Telephone”.
How to use
If the extension you are walking in has 'One Call" as the Walk-Out mode, and you go ON-Hook before you
make the call, you will be 'Walked Out'. You must Walk-In again.
Technical Specifications
System Resources
System Resources Description Maximum Capacity
Total User Ports The Maximum Physical 324 516 60 120 240 24 24 48
Ports available
Optional Interfaces
Maximum Ports
System
Description ETERNITY ME ETERNITY GE ETERNITY PE
Resources
10S 16S 3S 6S 12S 3SS 3SP 6SP
The maximum number of ports supported by the GSM, SLT and DKP Cards may vary according to the type
of Power Supply used. Refer the following table for maximum ports supported with Universal Power
(PSUNI) and DC Power Supply in ETERNITY ME.
Number of ports supported in talk mode (OFF-Hook short loop) 250 200 175 150 128
according to loop current programmed
Built-In Interfaces
ETERNITY ME ETERNITY GE ETERNITY PE
System Resources Description
10S 16S 3S 6S 12S 3SS 3SP 6SP
DKP32 32 DKP Ports card to connect 32 Digital Key Phones /DSS Consoles
DKP16 16 DKP Ports card to connect 16 Digital Key Phones /DSS Consoles
DKP8 8 DKP Ports card to connect 8 Digital Key Phones /DSS Consoles
TWT8+SLT24 8 Two-Wire Trunk Ports and 24 SLT ports card to connect 8 Two-Wire Trunk Lines and
24 Single Line Telephones
SLT8-Magneto8 8 SLT ports and 8 Magneto ports to connect 8 Single Line Telephones and 8 Magneto
stations
TWT8-Magneto8 8 TWT ports and 8 Magneto ports to connect 8 Two-Wire Trunk Lines and 8 Magneto
stations
SLT8+MAG2+TWT2+L 8 SLT ports, 2 Magneto Ports, 2 Two-Wire Trunk ports, 2 Loop Dial ports, and 2 E&M
D2+ENM2 ports to connect 8 Single Line Telephones, 2 Magneto phones, 2 Two-Wire Trunk lines, 2
E&M ports
BRI8 8 BRI Ports card to connect 8 ISDN BRI Lines or ISDN Compatible Devices
BRI4 4 BRI Ports card to connect 4 ISDN BRI Lines or ISDN Compatible Devices
T1E1PRI Dual Dual Port T1/E1PRI Card with QSIG Support to connect 2 T1/E1 PRI Lines or
Compatible Devices
T1E1PRI Single Single Port T1/E1PRI Card with QSIG Support to connect 2 T1/E1 PRI Lines or
Compatible Devices
GSM/3G8 8 GSM/Ports card to insert 8 GSM SIM Cards for mobile Network Connectivity
GSM/3G4 4 GSM/Ports card to insert 4 GSM SIM Cards for mobile Network Connectivity
VoIP32 VoIP Card to connect to IP Network for making VoIP calls. Supports 32 VoIP Channels
VMS16 16 Port Voice Mail System with 512 Mailboxes to give mailbox facility to PBX users
DKP16 16 DKP Ports card to connect 16 Digital Key Phones /DSS Consoles
DKP8 8 DKP Ports card to connect 8 Digital Key Phones /DSS Consoles
DKP4+SLT16 4DKP Ports and 16 SLT Ports card to connect 4 Digital Key Phones and 16 Single Line
Telephones
TWT2+DKP2+SLT16 2 Two-Wire Trunk ports, 2 Digital Key Phone ports and 16 SLT ports to connect 2 Two-
Wire Trunk Lines, 2 Digital Key Phones/DSS Consoles and 16 Single Line Telephones
TWT8+SLT8 8 Two-Wire Trunk Ports and 8 SLT Ports card to connect 8 Two-Wire Trunk Lines and 8
Single Line Telephones
TWT4+SLT16 4 Two-Wire Trunk Ports and 16 SLT Ports card to connect 4 Two-Wire Trunk Lines and
16 Single Line Telephones
BRI4 4 BRI Ports card to connect 4 ISDN BRI Lines or ISDN Compatible Devices
T1E1PRI Single Single Port T1/E1PRI Card with QSIG Support to connect 1 T1/E1 PRI Line or
Compatible Device
GSM/3G 4 4 GSM/ Ports card to insert 4 GSM SIM Cards for mobile Network Connectivity
VoIP32 VoIP Card to connect to IP Network for making VoIP calls. Supports 32 VoIP Channels.
VMS16 16 Port Voice Mail System with 512 Mailboxes to give mailbox facility to PBX users
DKP8 8 DKP Ports card to connect 8 Digital Key Phones /DSS Consoles
DKP4 + SLT4 4 DKP Ports and 4 SLT Ports card to connect 4 Digital Key Phones and 16 Single Line
Telephones
TWT4 + SLT4 4 Two-wire Trunk ports and 4 SLT (Analog phone) card
TWT4+ DKP4 4 Two-Wire Trunk Ports and 4 DKP Ports card to connect 4 Two-Wire Trunk Lines and 4
Digital Key Phones/DSS Consoles
TWT2+ DKP2+SLT4 2 Two-Wire Trunk Ports, 2 DKP Ports and 4 SLT Ports card to connect 2 Two-Wire
Trunks, 2 Digital Key Phones/DSS Consoles, 4 Single Line Telephones
BRI2* 2 BRI Ports card to connect 2 ISDN BRI Lines or ISDN Compatible Devices
T1E1PRI Single* Single Port T1/E1PRI Card with QSIG Support to connect 1 T1/E1 PRI Line or
Compatible Device
GSM/3G 4 4 GSM Ports card to insert 4 GSM SIM Cards for mobile Network Connectivity
VoIP16 VoIP Card to connect to IP Network for making VoIP calls. Supports 16 VoIP Channels
per Card
VMS 16 Port Voice Mail System with 512 Mailboxes to give mailbox facility to PBX users
Optional Items
EON42, EON48 Digital Key Phone
Technology
Type of Switching PCM/TDM, Digital Switching, 100% Non-Blocking
Slots Universal
REN 3
CLI Reception DTMF, FSK ITU-T V.23 and FSK Bellcore 202
CLI Reception DTMF, FSK ITU-T V.23 and FSK Bellcore 202
ISDN BRI
Channels 2B+D
AT&T 4ESS, DMS-100, ETSI NET3, ITU-T Q.921, ITU-T Q.931, NTT INS64, US Ni1
Switch Variant
(National ISDN1), France Vnx
ISDN PRI
Channels 2B+D and 30B+D
AT&T 2ESS, AT&T 5ESS, DMS-100, ETSI NET5, ITU-T Q.921, ITU-T Q.931, NTT
Switch Variant
INS64, US NI2 (National ISDN 2), QSIC ECMA, France VN
E1 CAS
Bit Rate 2048 kbps +/-50 ppm
T1 RBS
Bit Rate 1544 kbps +/- 50 ppm
FXS Loop Start, FXO Loop Start, FXS Ground Start, FXO Ground Start, E&M
Line Signaling
(Immediate, Wink Start, Wink Start FGD)
GSM Trunks
GSM Band (MHz) Quadband GSM850, EGSM900, DSC1800, PCS1900
External Antenna One Antenna per 4 GSM Ports,1.8/3.0dBi, 50, SMA (male) Connector, Omni-
directional with cable of 3 meters length
VoIP
VoIP Protocols SIP V2, SDP, RTP, RFC 2833
SIP 4 to 32 SIP Trunks (depending on model), Outbound Proxy Support, Display Name, User
Name, Password, URL, Proxy URL, Register URL, Register Interval
Voice Dynamic Jitter Buffer (Adaptive), Comfort Noise Generation and Voice Activity Detection
LED Indications 1-LED for System Status and 1-LED for Registrar Status
E&M Trunks
Type Type IV and Type V
Auxiliary Ports
Analog Input Port 0.7Vmrs, Isolated, Push-type connector
Digital Output Port VDC Max.= 60VDC, IDC Max.= 0.15A, Push-type connector
Power Supply
Option 1: Mains 90-265VAC, 47-63Hz
Supply Input
Option 2: 48VDC +/-20%
Mechanical
Dimensions (W x H x D) ETERNITY ME10S 48.20 x 33.00 x 29.90 cm (19.00 x 12.90 x 11.70 Inches)
Installation ETERNITY ME10S Wall Mount, Table Top, 19" Rack (Optional)
Compliance
EMI/EMC
FCC
LVD 73/23/EEC
EMC 89/336/EEC
Verify contents of the package shipped to you with the contents listed below. If any of the items is missing or
damaged, contact your Dealer/Reseller.
ETERNITY ME 10S/16S
1 ETERNITY MEa 1
5. Mounting Template 1
6. User Card 4
7. Quick Start 1
a. Factory fitted with the Power Card, Master Card and Switch Card.
If your model is ETERNITY 10SR with the Redundancy Option, there will be 2 Power Cards, 2 Master
Cards and 2 Switch Cards factory fitted.
ETERNITY ME Cards
ETERNITY ME PSUNI
ETERNITY ME Master
2. Cable (RJ45) 1
2. RJ45 Cable 1
3. RJ45 Cable 1
a. Factory fitted.
ETERNITY GE
1 ETERNITY GEa 1
4. Mounting Template 1
5. User Card 4
6. Quick Start 1
2. RJ45 Cable 1
3. RJ45 Cable 1
a. Factory fitted.
ETERNITY PE
1 ETERNITY PEa 1
5. Mounting Template 1
6. User Card 4
7. Quick Start 1
2. RJ45 Cable 1
2. RJ45 Cable 1
2. RJ45 Cable 1
3. RJ45 Cable 1
a. Factory fitted.
EON42
2. Handset 1
EON48
2. Handset 1
5. Foot Stand 1
DSS64
1. DSS64 Console 1
2. Cable RJ45 1
DSS72
1. DSS72 Console 1
2. Cable RJ45 1
3. Foot stand 1
EONSOFT
1. EONSOFT Dongle 1
3. Communication Cable 1
4. Screw m7/30 2
5. Screw Grip 2
6. Mounting Template 1
7. EONSOFT CD 1
PPM4
1. Matrix PPM4 1
4. Mounting Template 1
The Earth (Ground) is the most important safety procedure to prevent electrical shocks and fires. It protects from
lightning strikes, electrical transients, static discharges, electromagnetic interference and electrical hazards.
A proper earth must be in place to protect people and the system. The following explanation shows how a perfect
electrical earth can save lives.
P
Vs Vc
A.C. Input Electrical Parts of the Gadget
N Z1
Z2
Enclosure of the Gadget
Earth
This formula implies that if the impedance between the Chassis and the Earth is reduced to 0 then the Voltage on
the Chassis, i.e. VC, would be Zero and hence any person touching the enclosure will not get an electric shock.
Hence Z2 should be made Zero.
It is recommended that you provide a dedicated earth for the PBX/any other telecom equipment. This dedicated
earth is called the Telecom Earth (Ground).
Providing a separate Telecom Earth to the telecom equipment eliminates the possibility of any back-voltage on the
earth.
CO Line CO Line
System
Lightning
Protective Earth
Protectors
Terminal
Telecom Earth
Exchange Room
• Get a copper plate of size 1.5 feet x 1.5 feet x 0.25 feet.
• Connect a copper strip of size 1-inch wide, 3 mm thick and 6 feet length at the center of the copper plate
by welding or nuts and bolts.
• Insert a G.I. pipe onto the copper strip till it reaches the copper plate.
• Place this set up into the pit. Make sure that at least 4 inches of the G.I. pipe is above the ground level.
• Fill the bottom of the pit with a 1-inch layer of charcoal and salt in the proportion of 3:1 (3 parts charcoal, 1
part salt) and then cover with the soil.
• Connect a bare 14 SWG copper wire (double) on the top of the copper strip and run it to the exchange
room and connect it on the bus bar.
• The Bus bar is a copper strip, 4 inches long with 6 screws and nuts mounted on it. It has to be fixed on the
wall in the exchange room.
• The earth wire of the Primary Protection Modules (PPM) should be connected to this Bus bar.
Abbreviated Dialing
Account Code
Alarms
Auto Redial
Auto Redial 17
Barge-In
Barge-In 4
Call Chaining
Call Forward
Call Forward-If Busy-All Calls to External Number 132-Trunk Access Code-Dest. Number-#*
Call Forward-If No Reply-All Calls to External Number 133-Trunk Access Code-Dest. Number-#*
Call Forward-Department Group - Busy/No Reply 1175-Department Group Number (Access Code)-4-
Destination Number
Call Hold
Call Park
Call Pick Up
Call Transfer
Conference 3-Party
Conference-Unsupervised Flash-#
Conference Dial-In
Conference Multiparty
Department Call
Do Not Disturb
DND Override 4
Door Phone
Set Door Phone to route calls to an extension 1172-Access Code of Door Phone-1
Set Door Phone to route calls to an external number 1172-Access Code of Door Phone-2
Dynamic Lock
Emergency Conference
Emergency Dialing
Floor Service
Follow Me
Forced Answer
Forced Answer 5
Hot Desking
Hotline
Interrupt Request
Interrupt Request 3
Maid-In
Meet Me Paging
Message Wait
Mini Bar
Mute
Mute 1052
Operator
Call to Operator 9
Paging
Presence
Raid
Raid 5
RCOC
Reminder
Room Monitor
SA Command 1072
Exit SE Mode 00
Reserve a Trunk 6
User Absent/Present
User Password
Voice Help
To Walk-Out 111-0
Call Phases
Feature Access
Feature Name Matured Matured
Number Code Dial Routing Blocked Placed
2-Way 3-way
- 039 1072-025 Y Y
- 116 --
- 117 --
- 127 1072-182 Y Y
Abbreviated Dialing
Access Codes
Account Codes
Program account name for the account code 4851-1-Account Code-Account Name
Auto Answer
To program Auto Call Back when Busy/No Reply in a 1302-1-COS Group-Feature Number-Code
CoS group
To assign the CoS group with Auto Call Back Busy/No 5502-1-Template Number-Feature Number-Code
Reply to a Station Basic Feature Template
To apply the Station Basic Feature Template with Auto 5503-1-SLT-Template Number
Call Back when Busy and No Reply, to an SLT
To apply the Station Basic Feature Template with Auto 5504-1-DKP-Template Number
Call Back when Busy and No Reply, to a DKP
To program Dialed Number String for the Index of the 4751-1-ANT Table No.-Index-Dialed Number String-
ANT Table #*
To program Substitute Number String for the Index of 4752-1-ANT Table No.-Index-Substitute Number
the ANT Table String -#*
To enable Automatic Number Translation Flag on 6702-1-OG Trunk Bundle Number-Feature Number-
OGTB Code
To assign an Automatic Number Translation Table 6702-1-OG Trunk Bundle Number-Feature Number-
Code
Barge-In
BITE
To 5366-1-Slot Number
Call Budget
To program Call Budget Reset Mode for TWT 3304-1-TWT-Call Budget Reset Mode
To program Call Budget Reset Mode for SIP 7736-1-SIP-Call Budget Reset Mode
To program Call Budget Reset Mode for Mobile trunk 8022-1-Mobile-Call Budget Reset Mode
port
To program Call Budget Reset Mode for T1E1 6138-1-T1E1-Call Budget Reset Mode
Program the unit charge for first unit when 16 KHz 2600-Unit Charge for First Unit
metering is used
Program the unit charge for additional unit when 16 2601-Unit Charge for Additional Unit
KHz metering is used
Program duration of first unit for a pulse rate type on 2607-Pulse Rate Type-Time Zone-Duration of First
normal days Unit
Program duration of additional unit for a pulse rate 2608-Pulse Rate Type-Time Zone-Duration of
type on normal days Additional Unit
Program cost of first unit for a pulse rate type on 2609-Pulse Rate Type-Time Zone-Cost of First Unit
normal days
Program duration of first unit for a pulse rate type on 2612-Pulse Rate Type-Time Zone-Duration of First
holidays Unit
Program duration of additional unit for a pulse rate 2613-Pulse Rate Type-Time Zone-Duration of
type on holidays Additional Unit
Program cost of first unit for a pulse rate type on 2614-Pulse Rate Type-Time Zone-Cost of First Unit
holidays
Program cost of additional unit for a pulse rate type on 2615-Pulse Rate Type-Time Zone-Cost of
holidays Additional Unit
To program Pulse Rate Type for Pulse Rate Option of 2621-Area Code Index-Pulse Rate Option-Pulse
area code index Rate Type
To program pulse rate for an area code 2621-Area Code Index-Pulse Rate Type
To program ignore digit count when SP_SP LCR is 2623-Area Code Index-Ignore Digit Count
used
Call Hold
Call Logs
Call Park
Call Pick Up
To assign call pickup group for ISDN Terminal 3903-1-ISDN Terminal-Call Pickup Group
To clear an entry of the White List table 7817-1-VoIP Ethernet Port-Table Number Index
To clear all entries of the White List table of a VoIP 7817-1-VoIP Ethernet Port-#*
Ethernet Port
Call Taping
To enable Tape calls coming without CLI Flag in a 5602-1-Template Number-Feature Number-Code
Station Advanced Feature Template
Call Transfer
To assign landing destination for the incoming number 4103-Index-Port Type-Port Number
Clock Synchronization
To program strip digit count for a route 4504-1-Route Index-Strip Digit Count
To program maximum dialed digits to select router for 4506-1-Route Index-Maximum Dialed Digits
a route code
Communication Ports
To program Dynamic DNS - Retry Trial Count 2129 - DDNS Retry Count
To assign a Key Map for a DKP Port 1221-1-DKP-DKP Key Template Number
To assign the DKP Port to a Call Pick-Up Group 3902-1-DKP-Call Pickup Group
To select Ring Delay Timer for a DKP Port 1205-1-DKP-Ring Delay Timer
To set the Ringer Auto Acknowledge Timer 1207-1-DKP-Ringer Auto Acknowledge Timer
To select Destination for 'Play Ring ON' for a DKP Port 1220-1-DKP-Ring Destination
To set Handset Transmit (Tx) Volume Level for a DKP 1208-1-DKP-Handset MIC Volume Level
To set Handset Receive (Rx) Volume Level for a DKP 1209-1-DKP-Handset Speaker Volume Level
To set Headset Transmit (Tx) Transmit Volume Level 1222-1-DKP-Headset MIC Volume Level
for a DKP
To set Headset Receive (Rx) Volume Level for a DKP 1223-1-DKP-Headset Speaker Volume Level
To set Hands-free Transmit (Tx) Volume Level for a 1210-1-DKP-Speaker Phone MIC Volume Level
DKP
To set Hands-free Receive (Rx) Volume Level for a 1211-1-DKP-Speaker Phone Speaker Volume Level
DKP
To set Key Click Volume Level for a DKP 1212-1-DKP-Key Click Volume
To set Auto Answer Timer (sec) for a DKP 1215-1-DKP-Auto Call Answer Timer
To change LCD Backlight OFF Timer of a DKP 1219-1-DKP-LCD Backlight OFF Timer
To assign an ISDN Terminal to a Call Pick-Up Group 3903-1-ISDN Terminal-Call Pick-Up Group
Configuring Region
Configuring Operator
Configuring Trunks
To change the default value of a TWT Hardware 5902-1-TWT Hardware Template Number-Feature
Parameter in a Template Number-Code
To change the default value of a Trunk Feature 5802-1-Trunk Feature Template Number-Feature
Parameter in a Template Number-Code
To assign a Trunk Feature Template to a TWT Trunk 5803-1-TWT-Trunk Feature Template Number
To assign a Trunk Feature Template to a BRI Trunk 5804-1-BRI-Trunk Feature Template Number
To assign a Trunk Feature Template to an E&M Trunk 5805-1-E&M-Trunk Feature Template Number
To assign a Trunk Feature Template to a T1E1 Trunk 5806-1-T1E1-Trunk Feature Template Number
To assign a Trunk Feature Template to a SIP Trunk 5808-1-SIP-Trunk Feature Template Number
To change the default values of a SIP Hardware 7806-1-SIP Hardware Template Number-Parameter
Parameter in a Template Number-Code
To default the values of a SIP Hardware Template 7805-1-SIP Hardware Template Number
To apply SIP Hardware Template to a SIP Trunk 7808-1-SIP-SIP Hardware Template Number
To apply SIP Hardware Template to a SIP Extension 7807-1-SIP Extension-SIP Hardware Template
Number
To assign Hardware Slot and Port to the TWT Port 1104-TWT-Slot-Port offset on the card
To assign a Trunk Feature Template to a TWT Port 5803-1-TWT-Trunk Feature Template Number
To select the frequency Band for the Mobile Port 8009-1-Mobile Port Number-Mobile Frequency
Band Code
To assign SIM PIN to the Mobile Port 8006-1-Mobile Port Number-SIM PIN-#*
To assign a Trunk Feature Template to the Mobile Port 5807-1-Mobile Port Number-Template Number
To configure Subnet Mask for the LAN Port 7822-1-VoIP Port-Subnet Mask
To configure Cloned MAC Address on WAN Port 7774-1-VoIP Port-Cloned MAC Address
To select the Connection Type for the VoIP Ethernet 7751-1-VoIP Port-Connection Type
Port
To configure the Service Name for a PPPoE 7787-1-VoIP Port-PPPoE Service Name
Connection Type
To configure IP Address for VoIP Port having 'Static' as 7754-1-VoIP Port-IP Address
Connection Type
To configure Subnet Mask for the VoIP Port having 7755-1-VoIP Port-Subnet Mask
'Static' as Connection Type
To configure Gateway IP Address for the VoIP Port 7756-1-VoIP Port-Gateway Address
having 'Static' as Connection Type
To program User ID for Dynamic DNS 7825-1-VoIP Port-User ID for Dynamic DNS
To program password for Dynamic DNS 7826-1-VoIP Port-Password for Dynamic DNS
To program Host Name for Dynamic DNS 7827-1-VoIP Port-Host Name for Dynamic DNS
To program Retry Trials for Dynamic DNS 7828-1-VoIP Port-Retry Trial for Dynamic DNS
To select SIP DiffServe/ToS as Quality of Service for a 7765-1-VoIP Port-SIP QoS Level
VoIP Port
To select RTP DiffServe/ToS as Quality of Service for 7767-1-VoIP Port-RTP QoS Level
a VoIP Port
To define SIP UDP Port for a VoIP Port 7768-1-VoIP Port-SIP UDP Port
To define SIP TCP Port for a VoIP Port 7784-1-VoIP Port-SIP TCP Port
To enable/disable UDP NAT Keep Alive for a VoIP Port 7761-1-VoIP Port-UDP NAT Keep Alive
To select the Type of Message for UDP NAT Keep 7778-1-VoIP Port-UDP NAT Keep Alive Message
Alive for a VoIP Port Type
To enable/disable TCP NAT Keep Alive for a VoIP Port 7785-1-VoIP Port -TCP NAT Keep Alive
To configure the SIP General Request Timer for a 7779-1-VoIP Port-SIP General Request Timer
VoIP Ethernet Port
To configure the Authentication Password for the SIP 7710-1-SIP-Authentication User Password
trunk
To enable/disable Outbound Proxy for the SIP trunk 7715-1-SIP-Outbound Proxy Status
To configure Server Address of Outbound Proxy for 7716-1-SIP-Outbound Proxy Server Address
the SIP trunk
To define Outbound Proxy Server's Listening Port 7717-1-SIP-Outbound Proxy Server Port
To select the OG Vocoder Preference for a SIP trunk 7712-1-SIP-Preference Index-Preference Vocoder
To assign a Station Basic Feature Template to a SIP 5508-1-SIP-Station Basic Feature Template
trunk Number
To program Fax Data Gain for SIP Trunk in case of 7887-1-SIP-Fax Data Gain
SIP to Digital Trunk Call
To program Fax Bypass Gain for SIP Trunk in case of 7889-1-SIP-Fax Bypass Gain
SIP to Digital Trunk Call
To program Fax Data Gain for SIP Trunk in case of 7888-1-SIP-Fax Data Gain
SIP to SLT Call
To program Fax Bypass Gain for SIP Trunk in case of 7890-1-SIP-Fax Bypass Gain
SIP to SLT Calls
To program Echo Cancellation Tail Length for FXS and 7748-1-SIP-Tail Length
Digital Trunks
To configure the DTMF ON Time for the SIP trunk 7725-1-SIP-DTMF ON Time
To configure the DTMF Inter-Digit Pause Timer for the 7726-1-SIP-DTMF Inter Digit Pause Timer
SIP trunk
To configure DTMF String for Gateway Application- 7728-1-SIP-Gateway Application DTMF String
Answer Signaling on the SIP trunk
To assign a Station Basic Feature Template to a SIP 5510-1-SIP Extension-Station Basic Feature
Extension Template
To assign a SIP Extension to a Call Pick-Up Group 3904-1-SIP Extension-Call Pick Up Group
To assign a VoIP Port Number for SIP Extension 7871-1-SIP Extension-VoIP Port
To define the Call Appearance for a SIP Extension 7876-1-SIP Extension-Call Appearance
To enable/disable authenticate Voice Mail Subscription 7882-1-SIP Extension-Voice Mail Subscription Flag
flag
To assign a Station Basic Feature Template to a SIP 5510-1-SIP Extension-Station Basic Feature
Extension Template
To assign a SIP Extension to a Call Pick-Up Group 3904-1-SIP Extension-Call Pick Up Group
To assign Slot-Port Assignment to a Magneto Port 1110-Magneto Trunk-Slot-Port offset on the Card
To assign the SLT Hardware Template for LD trunk 5704-1-LD-SLT Hardware Template
port
To program the Call Cost Calculation Time Schedule- 3951-1-LD Trunk-Start Time
T3-Start Time for LD trunk port
Configuring LCR
To program Time Zone at a Time Zone index 3402-Time Zone Index-Start Time-End Time
To define Time Zone for Time+Number-based LCR 3421-Time Zone Index-Start Time-End Time
To program Cost Factor (Service Provider preference) 3423-Number Index-Time Zone Index-CF1-CF2-
for the each Number and Time Zone CF3-CF4
To program Area Code in the Area Code Table 2620-Area Code Index-Area Code-#*
To clear an Area Code in the Area Code Table 2620-Area Code Index-#*
To program Ignore Digit Count for an Area Code 2623-Area Code Index-Ignore Digit Count
To program OG Trunk Bundle Group (OTBG) for an 3117-Index-OG Trunk Bundle Group
emergency number
Conflict Dialing
Conversation Recording
Customer Name
To program the end time of DST when manual 1012-Date-Month-Current Time-Delay Time
selected
To program the feature in a DDI Routing Table 6322-1-DDI Routing Table ID-Parameter Number-
Value
To enable/disable debug for the VoIP Ethernet Port 7791-1-VoIP Ethernet Port-Debug
To program Syslog Server Address on which Debug 7792-1-VoIP Ethernet Port-Syslog Server Address
parameter is to be sent for the VoIP Ethernet Port
To program Server Port Address on which Debug 7793-1-VoIP Ethernet Port-Server Port Address
parameter is to be sent for the VoIP Ethernet Port
To enable debug level for the VoIP Ethernet Port 7794-1-VoIP Ethernet Port-Index-Code
Default Settings
Department Call
To program the time for which each station in the 6503-1-Routing Group-Destination Index-Ring
group should ring Timer
To assign routing group to the department group 2001-1-Department Group Index-Routing Group
To clear the routing group assigned to the department 2001-1-Department Group Index-00
group
To program the access code for a department number 3113-1-Department Group Index-Access Code
To default the access code for a department group 3163-1-Department Group Index
To clear the access code for a department number 3113-1-Department Group Index-#*
Digest Authentication
To program the minimum instigation time for a DIP 4903-1-DIP-Minimum Instigation Time
To program the contact type for the DOP 5001-1-DOP-Normal Contact Type
To program Port Type and Port Number for DISA 4112- Index-Port Type-Port Number
Application
Distinctive Rings
Door Phone
To assign call routing mode for Door Phone 3222-1-Door Phone-Call Routing Mode
To program Scheduled - Door Phone Call Routing for 3223-1-Door Phone-Time Zone-Call Routing Type
each Time Zone
To program Scheduled - Door Phone Call Routing on 3224-1-Door Phone-Time Zone-Routing Group
Routing Group
To program Manual-Door Phone Call Routing Mode 3226-1-Door Phone-Call Routing Mode
To program Manual - Routing Group for Door Phone 3227-1-Door Phone-Routing Group
calls
To program the Door Phone Ring Timer 3230-1-Door Phone-Door Phone Ring Timer
To assign DOP (Door Opener) for the Door Phone 3231-1-Door Phone-DOP
To program the Open Door Timer 3232-1-Door Phone-Open Door for Time
Emergency Dialing
Flexible Numbers
Floor Service
To program the Ring Timer for the routing group 6503-1-Routing Group-Destination Index-Ring
Timer
To program the Continuous Ring Flag for the routing 6504-1-Routing Group-Destination Index-Flag
group
IC Reference Table
Interrupt Request
To program the BRI ISDN switch variant of the BRI 6203-1-BRI-BRI ISDN Switch Variant
port
To program DTMF Inter digit Pause Timer 6211-1-BRI- DTMF Inter digit Pause Time
To program a called party TON for the BRI 6223-1-BRI-Called Party TON
To program a called party NPI for the BRI 6224-1-BRI-Called Party NPI
Logical Partition
Message Wait
To program the message wait ring timer 4404-Message Wait Ring Timer
To program the message wait ring interval timer 4405-Message Wait Ring Interval Timer
Music on Hold
Number List
OFF-Hook Alert
OG Trunk Bundle
To program the feature in OG trunk bundle 6702-1-OG Trunk Bundle Number-Port Type-Port
Number-Code
To set default values for OG trunk bundle 6701-1-OG Trunk Bundle Number
To program the desirable access code for a trunk 3112-1-OGTBG Index-Access Code-#*
access index
Paging
Presence
Peer-to-Peer Calling
Priority
RCOC
Reminder
Routing Group
To program OG Trunk Bundle Group for External 5208-OG Trunk Bundle Group
Numbers
To program Delay Response Timer for Security Alarm 5209-Delay Response Timer
To program Call Attempt Interval for External Number 5210-Call Attempt Interval
To program property code string for property code 8209-Property Code String
To program date fill flag for date field 8257-Date Fill Flag
To program time fill flag for time field 8258-Time Fill Flag
To enable/disable the Filler char. flag for Answer 8259-Filler Character Flag for Answer Duration
Duration
To enable/disable the Filler character flag for Hold 8260-Filler Character Flag for Hold Duration
Duration
To enable/disable the Filler char. flag for Speech 8261-Filler Character Flag for Speech Duration
Duration
To set Data Transfer Retry Count (No Response) 8308-Data Transfer Retry Count
To set Data Transfer Retry Time (No Response) 8309-Data Transfer Retry Time
To set Data Transfer Retry Count (Negative 8310-Data Transfer Retry Count
Response)
To set Data Transfer Retry Time (Negative Response) 8311-Data Transfer Retry Time
To program property code string for property code 8109-Property Code String
To program date fill flag for date field 8170-Date Fill Flag
To program time fill flag for time field 8171-Time Fill Flag
To program duration fill flag for duration field 8172-Duration Fill Flag
To program amount fill flag for amount field 8173-Amount Fill Flag
To program column position for call type indicator field 8146-Column Position
To program field length for call type indicator field 8147-Field Length
To program number string for call type indicator field 8149-Number Index-1-Number String
To program text string for call type indicator field 8149-Number Index-2-Text String
System Debug
To enable or disable PCM capture - Debug 2124-Slot Number-Hardware Port Offset- Code
System Parameters
To program the on/off time of the ETERNITY ME Card 5308-ON Timer-OFF Timer
Master buzzer
To program Call Proceeding Tone Type for multi-stage 5311-Call Proceeding Tone Type
dialing
System Security
T1 Maintenance
To program the T1 FDL protocol for a T1E1 port 6165-1-T1E1-T1 FDL Protocol
T1 RBS Parameters
To program the T1 line signaling variants for the T1E1 6181-1-T1E1-T1 Line Signaling Variants
port
To program the T1 wink wait timer for T1E1 6183-1-T1E1-Wink Wait Timer
To program the T1 wait wink timer for T1E1 6184-1-T1E1-Wait Wink Timer
To program the T1 start delay duration for T1E1 6186-1-T1E1-Start Delay Duration
To program the T1 DTMF digit timer for T1E1 6187-1-T1E1-DTMF Digit Timer
To program the T1 DTMF inter digit timer for T1E1 6188-1-T1E1-DTMF Inter Digit Timer
To program digital pulse dial ratio for the T1E1 port 6163-1-T1E1-Code
T1E1 Trunks
To program the line coding mechanism for the T1E1 6103-1-T1E1-Line Coding
port
To program DTMF Inter digit Pause Timer 6118-1-T1E1-DTMF Inter Digit Pause Timer
To program E1 line signaling variant for the T1E1 6152-1-T1E1-E1 Line Signaling Variant
To program E1 register signaling variant for the T1E1 6153-1-T1E1-E1 Register Signaling Variant
To program customer pulse width word1 for T1E1 T1 6172-1-T1E1-Customer Pulse Width Word 1
signaling
To program customer pulse width word2 for T1E1 T1 6173-1-T1E1-Customer Pulse Width Word 2
signaling
To program customer pulse width word3 for T1E1 T1 6174-1-T1E1-Customer Pulse Width Word 3
signaling
To program customer pulse width word4 for T1E1 T1 6175-1-T1E1-Customer Pulse Width Word 4
signaling
To program customer pulse width word1 for T1E1 E1 6156-1-T1E1-Customer Pulse Width Word 1
signaling
To program customer pulse width word2 for T1E1 E1 6157-1-T1E1-Customer Pulse Width Word 2
signaling
To program customer pulse width word3 for T1E1 E1 6158-1-T1E1-Customer Pulse Width Word 3
signaling
To program customer pulse width word4 for T1E1 E1 6159-1-T1E1-Customer Pulse Width Word 4
signaling
To program the ISDN PRI switch variant 6107-1-T1E1-ISDN PRI Switch Variant
To program forward tone maximum OFF timer 7102-1-T1E1-Forward Tone Maximum OFF Timer
To program the maximum compelled cycle time 7103-1-T1E1-Maximum Compelled Cycle Time
To program pulse duration for pulsed signals 7104-1-T1E1-Pulse Duration for Pulsed Signals
To program the pulsed signal maximum wait timer 7105-1-T1E1-Pulsed Signal Maximum Wait Timer
To program first forward tone wait timer 7106-1-T1E1-First Forward Tone Wait Timer
To program to set DNIS END type (outbound) for T1E1 7108-1-T1E1-End of DNIS
To program the positive response to Is ANI available 7112-1-T1E1-Positive Response to Is ANI Available
(outbound)
To program the end of ANI with presentation allowed 7114-1-T1E1-ANI End Tone with Presentation
(outbound) Allowed
To program the end of ANI with presentation restrict 7115-1-T1E1-ANI End Tone with Presentation
(outbound) Restrict
To program end tone presentation allowed (inbound) 7122-1-T1E1-ANI End Tone Presentation Allowed
To program end tone presentation restrict (inbound) 7123-1-T1E1-ANI End Tone Presentation Restrict
To program the send last but one digit 7132-1-T1E1-Send Last but One Digit
To program the send last but two digit 7134-1-T1E1-Send Last but Two Digit
To program the send last but three digit 7135-1-T1E1-Send Last but three Digit
To program the address complete, change over to 7136-1-T1E1-Address Complete, Change Over to
reception of group B signals Reception of Group B
To program the send calling party category and 7137-1-T1E1-Send Calling Party Category and
change to group C Change to Group C
To program the send caller party's category 7139-1-Send Caller Party Category
To program the send next ANI digit 7142-1-T1E1-Send Next ANI Digit
To program the send special information tone 7143-1-T1E1-Send Special Information Tone
To program the send special information tone and 7144-1-T1E1-Send Special Information Tone and
setup speech condition Setup Speech Condition
To program the subscriber line free, charge 7146-1-T1E1-Subscriber Line Free, Charge
To program the subscriber line free, no charge 7147-1-T1E1-Subscriber Line Free, No Charge
To program the subscriber's line out of order 7150-1-T1E1-Subscriber's Line Out of Order
To program the call rejected, no indication of cause 7151-1-T1E1-Reject Call due to R2MF Tone
To program the send next ANI digit (group C) 7154-1-T1E1-Send Next Digit
To program the request transition to group A and 7155-1-T1E1-Request Transition to Group A and
restart from first DNIS Restart from First DNIS
To program the tone for request transition back to 7158-1-T1E1-Request Transition Back to Group A,
group A and send next DNIS signal and Send Next DNIS
To program the tone for request transition back to 7159-1-T1E1-Request Transition Back to Group A,
group A and restart the last DNIS signal and Restart the Last DNIS
To program the invert/don't invert bit A for the T1E1 7162-1-T1E1-Invert Bit A
To program the invert/don't invert bit B for the T1E1 7163-1-T1E1-Invert Bit B
To program the invert/don't invert bit C for the T1E1 7164-1-T1E1-Invert Bit C
To program the invert/don't invert bit D for the T1E1 7165-1-T1E1-Invert Bit D
To program the E1 metering bit for the T1E1 port 7166-1-T1E1-E1 Metering Bit
To program the metering pulse minimum timer for the 7167-1-T1E1-E1 Metering Pulse Minimum Timer
T1E1
To program the clear back signal for the T1E1 7168-1-T1E1-Clear Back Signal
To program line seizure acknowledge wait timer 7170-1-T1E1-Line Seizure Acknowledge Wait Timer
To program to invert/don't invert Bit A for the T1E1 port 7162-1-T1E1-Invert Bit A
To program to invert/don't invert Bit B for the T1E1 port 7163-1-T1E1-Invert Bit B
To program to invert/don't invert Bit C for the T1E1 port 7164-1-T1E1-Invert Bit C
To program to invert/don't invert Bit D for the T1E1 port 7165-1-T1E1-Invert Bit D
Time Tables
Toll Control
Trunk Reservation
To program the access code for a VMS group 3114-1-VMS Group Index-Access Code-#*
To default the access code for a VMS group 3164-1-VMS Group Index
To clear the access code for a VMS group 3114-1-VMS Group Index-#*
To know 'IP Address: Web Server Port' of VMS Card 2131-Slot Number
To de-assign the voice module from a voice message 2505-Voice Message Application Number-00
application
• All servicing to be undertaken ONLY by qualified service personnel. There are no user serviceable parts
inside the unit.
• Always switch off "MAINS" and "BATTERY" marked switches of the system before opening the system and
remove power cable from Mains plug, to avoid risk of electric shock.
AC Mains:
• Check the Mains Voltage.
• Check the Mains Fuse (6Amp Slow Blow, glass fuse provided in AC Mains socket of ETERNITY ME Card
10SAC/16SAC).
• Check for loose connection of PT3 connector (connecting AC Mains socket to ETERNITY ME Card
10SAC/16SAC).
• If FCBC is used, check battery voltage interfaced with FCBC, it has to be 43-56V.
• Check wiring.
• Check the Time programmed in the PBX. This is a time sensitive feature.
• Please check whether the Station where you are checking CLI function is programmed as CLI Phone.
• Please check for the polarity. Refer to the figure shown in topic “Digital Output Port (DOP)”.
• Please check SIP trunks are programmed correctly in the Routing Group.
AVD Alternate Voice data signaling (Also called clear channel, out-of-band signaling)
BH Break Hour
BI Barge-In
CD Carrier Detect
CI Call Incoming
CO Central Office
E1 E-Carrier1 (30B+D)
ENQ Enquiry
FM Frequency Modulation
GND Ground
IC Incoming call
IP Internet Protocol
IR Interrupt
LA Left Align
LE Local Exchange
MS Mobile Station
OG Outgoing
PC Personal Computer
PS Power Supply
QSIG Q-Signaling
RA Right Align
RF Radio Frequency
RI Ring Indicator
SA System Administrator
SE System Engineer
SP Service Provider
T1 T-Carrier (23B+D)
TA Terminal Adaptor
WH Working Hour
Matrix Comsec Pvt. Ltd. (Matrix) warrants to its consumer purchaser any of its products to be free of defects in
material, workmanship and performance for a period of 15 months from date of manufacturing or 12 months from
the date of installation which ever is earlier.
During this warranty period, Matrix will at its option, repair or replace the product at no additional charge if the
product is found to have manufacturing defect. Any replacement product or part(s) may be furnished on an
exchange basis, which shall be new or like-new, provided that it has functionality at least equal to that of the
product, being replaced. All replacement parts and products will be the property of Matrix. Parts repaired or
replaced will be under warranty throughout the remainder of the original warranty period only.
1. Products that have been subjected to abuse, accident, natural disaster, misuse, modification, tampering,
faulty installation, lack of reasonable care, repair or service in any way that is not contemplated in the
documentation for the product or if the model or serial number has been altered, tampered with, defaced or
removed.
2. Products which have been damaged by lightning storms, water or power surges or which have been
neglected, altered, used for a purpose other than the one for which they were manufactured, repaired by
customer or any party without Matrix’s written authorization or used in any manner inconsistent with
Matrix’s instructions.
4. Products damaged due to operation of product outside the products’ specifications or use without
designated protections.
Warranty Card:
• When the product is installed, please return the warranty card with:
• Matrix assumes that the customer agrees with the warranty terms even when the warranty card is not
signed and returned as suggested.
The Purchaser shall have to bear shipping charges for sending product to Matrix for testing/rectification. The
product shall be shipped to the Purchaser at no-charge if the material is found to be under warranty. The Purchaser
shall have to either insure the product or assume liability for loss or damage during transit.
Matrix reserves the right to waive off or make any changes in its warranty policy without giving any notice.
In no event will Matrix be liable for any damages including lost profits, lost business, lost savings, downtime or
delay, labor, repair or material cost, injury to person, property or other incidental or consequential damages arising
out of use of or inability to use such product, even if Matrix has been advised of the possibility of such damages or
losses or for any claim by any other party.
Except for the obligations specifically set forth in this Warranty Policy Statement, in no event shall Matrix be liable
for any direct, indirect, special, incidental or consequential damages whether based on contract or any other legal
theory and where advised of the possibility of such damages.
Neither Matrix nor any of its distributors, dealers or sub-dealers makes any other warranty of any kind, whether
expressed or implied, with respect to Matrix products. Matrix and its distributors, dealers or sub-dealers specifically
disclaim the implied warranties of merchantability and fitness for a particular purpose.
This warranty is not transferable and applies only to the original consumer purchaser of the Product. Warranty shall
be void if the warranty card is not completed and registered with Matrix within 30 days of installation.
All legal course of action subjected to Vadodara (Gujarat, India) Jurisdiction only.
Customer Information-ACTA
Part 15:
Note: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to
Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference
when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate
radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful
interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful
interference in which case the user will be required to correct the interference at his/her own expense.
Part 68:
This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA (Administrative
Council for Terminal Attachments). On the bottom side of this equipment is a label that contains, among other
information, a product identifier in the format US: MTXMF01BETERNITY. If requested, this number must be
provided to the telephone company.
REN Number
The REN is used to determine the quantity of devices that may be connected to the telephone line. Excessive
RENs on the telephone line may result in devices not ringing in response to an incoming call. In most, but not all
areas, the sum of RENs should not exceed 5.0. To be certain of the number of devices that may be connected to a
line, as determined by the total RENs, contact the local telephone company. On the bottom of this equipment is a
label that contains, among other information, a product identifier in the format US: MTXMF01BETERNITY. The
digits represented by 01 are the ringer equivalence number (REN) without a decimal point (for example, 03 is a
REN of 0.3). If requested, this number must be provided to the telephone company.
If this equipment 'ETERNITY' causes harm to the telephone network, the telephone company will notify you in
advance that temporary discontinuance of service may be required. But if advance notice is not practical, the
telephone company will notify the customer as soon as possible. Also, you will be advised of your right to file a
complaint with the FCC if you believe it is necessary.
The telephone company may make changes in its facilities, equipment, operations or procedures that could affect
the operation of the equipment. If this happens, the telephone company will provide advance notice in order for you
to make necessary modifications to maintain uninterrupted service.
If trouble is experienced with this equipment 'ETERNITY', for repair or warranty information, please contact your
dealer. If the equipment is causing harm to the telephone network, the telephone company may request that you
disconnect the equipment until the problem is resolved.
The equipment cannot be used on public coin phone service provided by the telephone company. Connection to
party line service is subject to state tariffs. Contact the state public utility commission, public service commission or
corporation commission for information.
a. This equipment returns answer supervision to the public switched telephone network (PSTN) when DID
calls are:
b. This equipment returns answer supervision on all DID calls forwarded to the PSTN. Permissible
exceptions are:
• A call is unanswered
• A busy tone is received
• A reorder tone is received
Equal Access:
This equipment is capable of providing the end user equal access to the carrier of the user's choice. This
equipment is capable of providing users access to interstate providers of operator services through the use of
access codes. Modification of this equipment by call aggregators to block access dialing codes is a violation of the
Telephone Operator Consumers Act of 1990.
Electrical Safety:
Telephone companies report that electrical surges, typically lightning transients, are very destructive to customer
terminal equipment connected to AC power sources. This has been identified as a major nationwide problem.
However Matrix provides all protection against lightning transients in the equipment; the user must provide a
suitable surge arrestor while integrating the equipment with other networking equipments.
To ensure proper operation, this equipment must be installed according to the enclosed installation
instructions. To verify that the equipment is operating properly and can successfully report an alarm, this
equipment must be tested immediately after installation, and periodically thereafter, according to the
enclosed test instructions.
Any repairs or alterations made by the user to this equipment, or equipment malfunctions, may give the
telecommunications company cause to request the user to disconnect the equipment.
• The source of the open source software used in this product is available on CD, upon written request from:
R&D Team
Matrix Comsec Pvt Ltd
394, Makarpura GIDC,
Vadodara - 390 010
Gujarat
India.
Customer shall bear the shipping and handling charges.
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
License is intended to guarantee your freedom to share and change free
software--to make sure the software is free for all its users. This
General Public License applies to most of the Free Software
Foundation's software and to any other program whose authors commit to
using it. (Some other Free Software Foundation software is covered by
the GNU Lesser General Public License instead.) You can apply it to
your programs, too.
Also, for each author's protection and ours, we want to make certain
that everyone understands that there is no warranty for this free
software. If the software is modified by someone else and passed on, we
want its recipients to know that what they have is not the original, so
that any problems introduced by others will not reflect on the original
authors' reputations.
You may charge a fee for the physical act of transferring a copy, and
you may at your option offer warranty protection in exchange for a fee.
2. You may modify your copy or copies of the Program or any portion
b) You must cause any work that you distribute or publish, that in
whole or in part contains or is derived from the Program or any
part thereof, to be licensed as a whole at no charge to all third
parties under the terms of this License.
3. You may copy and distribute the Program (or a work based on it,
under Section 2) in object code or executable form under the terms of
Sections 1 and 2 above provided that you also do one of the following:
The source code for a work means the preferred form of the work for
making modifications to it. For an executable work, complete source
code means all the source code for all modules it contains, plus any
associated interface definition files, plus the scripts used to
control compilation and installation of the executable. However, as a
special exception, the source code distributed need not include
anything that is normally distributed (in either source or binary
form) with the major components (compiler, kernel, and so on) of the
operating system on which the executable runs, unless that component
itself accompanies the executable.
5. You are not required to accept this License, since you have not
signed it. However, nothing else grants you permission to modify or
distribute the Program or its derivative works. These actions are
prohibited by law if you do not accept this License. Therefore, by
modifying or distributing the Program (or any work based on the
Program), you indicate your acceptance of this License to do so, and
all its terms and conditions for copying, distributing or modifying
the Program or works based on it.
6. Each time you redistribute the Program (or any work based on the
Program), the recipient automatically receives a license from the
original licensor to copy, distribute or modify the Program subject to
these terms and conditions. You may not impose any further
restrictions on the recipients' exercise of the rights granted herein.
You are not responsible for enforcing compliance by third parties to
this License.
9. The Free Software Foundation may publish revised and/or new versions
of the General Public License from time to time. Such new versions will
be similar in spirit to the present version, but may differ in detail to
address new problems or concerns.
NO WARRANTY
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
You should have received a copy of the GNU General Public License along
with this program; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
Also add information on how to contact you by electronic and paper mail.
The hypothetical commands `show w' and `show c' should show the appropriate parts
of the General Public License. Of course, the commands you use may be called
something other than `show w' and `show c'; they could even be mouse-clicks or
menu items--whatever suits your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License.
I R
IC Reference Table 1252–1254 Raid 1417–1418
Installing ETERNITY 33–39 RCOC (Return Call to Original Caller) 1419–1423
Installing ETERNITY GE 165–251 Real Time Clock (RTC) 1424–1429
Installing ETERNITY ME 45–49 Reminder 1430–1438
Installing ETERNITY PE 253–325 Reminder - Personalized 1430
Internal Call Restriction 1259–1260 Reminder - Snooze 1432
Interrupt Request 1261–1262 Reminder - Voice-guided 1430
ISDN-BRI 1263–1284 Reminder Status 1436
ISDN-PRI 1285–1289 Remote Programming 1439–1440
Room Monitor 1441–1442
L Routing Group 1443–1446
Last Caller Recall 1294
Last Number Redial 1295 S
Least Cost Routing-Carrier Pre-Selection 1296–1297 Security Alarm and Reporting 1447–1455
License Management 1298–1309 Selective Port Access 1456–1457
Live Call Screening 1310–1312 Self Ring Test 1458
Live Call Supervision 1312 SIM Card Balance and Recharging 1459–1461
Logical Partition 1313–1316 Software Port and Hardware ID 1462–1466
Static Routing Table 1467–1471
M Station Message Detail Recording 1472
Meet Me Paging 1317–1318 Station Message Detail Recording-Online 1473–1488
Message Wait 1319 Station Message Detail Recording-Posting 1489–1530
Mobility Extension 1324–1330 Station Message Detail Recording-Report 1531–1544
Multi-Stage Dialing 1331–1334 Station Message Detail Recording-Storage 1545–1550
Music on Hold (MoH) 1335–1336 Station Name Pattern 1566
Mute 1342–1343 System Activity Log 1551–1556
System Activity Log Display 1557
System Debug 1084–1088
System Fault Log 1559–1564
System Fault Log Display 1558
System Parameters 1565–1583
System Security 1584
T
T1 Maintenance 1600–1607
T1 RBS Parameters 1660–1670
T1E1 Trunks 1608–1659
Time Tables 1671–1674
Time Zone Display 1675
Toll Control 1676–1691
traffic type 159, 247, 317
Trunk Access Group (TAG) 1692
Trunk Auto Answer 1693–1694
Trunk Landing Group (TLG) 1700–1702
Trunk Reservation 1703–1704
U
User Absent/Present 1705–1706
User Password 1707–1708
V
Video Call 1709
Virtual Station 1710
VLAN header 159, 247, 317
VLAN ID 159, 247, 317
VLAN/CoS Layer
CoS 159, 247, 317
RTP CoS 159, 247, 317
SIP CoS 159, 247, 317
VLAN ID 159, 247, 317
Voice Help 1711
Voice Mail Integration 1712–1715
Voice Message Applications 1716–1725
W
Walk-In Class of Service 1726–1728
MATRIX COMSEC PVT. LTD.
Corporate Office:
394-GIDC, Makarpura, Vadodara - 390010, India.
Tel.:+91 265 2630555, Fax: +91 265 2636598
E-mail: [email protected]
Factory:
39-GIDC, Waghodia - 391760, Dist. Vadodara, India.
Version 10, June 2011
Technical Support:
Tel.: +91 2668 263172/73, Fax: +91 2668 262631
E-mail: [email protected]
www.MatrixComSec.com