CCME System Admin
CCME System Admin
Americas Headquarters
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
USA
http://www.cisco.com
Tel: 408 526-4000
800 553-NETS (6387)
Fax: 408 527-0883
THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT
SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE
OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.
The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCB’s public
domain version of the UNIX operating system. All rights reserved. Copyright © 1981, Regents of the University of California.
NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED “AS IS” WITH
ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT
LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF
DEALING, USAGE, OR TRADE PRACTICE.
IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING,
WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO
OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
CCVP, the Cisco logo, and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn is a service mark of Cisco Systems,
Inc.; and Access Registrar, Aironet, BPX, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press,
Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Enterprise/Solver, EtherChannel, EtherFast, EtherSwitch, Fast Step, Follow Me Browsing,
FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, IP/TV, iQ Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, LightStream, Linksys,
MeetingPlace, MGX, Networking Academy, Network Registrar, Packet, PIX, ProConnect, ScriptShare, SMARTnet, StackWise, The Fastest Way to Increase Your Internet
Quotient, and TransPath are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a partnership relationship
between Cisco and any other company. (0705R)
Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the
document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.
FEATURE MAP 31
Contents 47
Additional References 52
Related Documents 52
Related Websites 54
MIBs 54
Obtaining Documentation, Obtaining Support, and Security Guidelines 54
Contents 55
Planning Worksheets 67
Contents 69
Contents 91
Contents 119
Contents 147
Contents 213
Contents 245
Contents 257
Contents 267
Contents 287
Contents 303
Contents 339
Contents 355
Prerequisites 355
Contents 387
Contents 461
Contents 467
Contents 485
Contents 499
Contents 563
Contents 639
Contents 647
Contents 689
Contents 709
Contents 717
Contents 729
Contents 737
Contents 745
Contents 753
Contents 763
Contents 771
Contents 785
Contents 797
Contents 819
Contents 829
Contents 847
Contents 867
Contents 881
Contents 893
Contents 919
Contents 943
Contents 959
Information About XML API 959
XML API Definition 959
XML API Provision Using IXI 960
How to Configure XML API 960
A C
Abbreviated Dialing Speed Dial 856 Call Blocking Based on Date and Time 467
Account Code Entry 831 Call Blocking Override 467
Ad Hoc Conferencing 647 Call Forwarding 500
Adding Directory Entries 689 Call Forwarding Support 499
After-Hours Call Blocking 467 Call Hold 819
After-Hours Toll Bar 467 Call Hunt 565
Agent Availability, Hunt Groups 575 Call Park 485
Analog Phones 147 Call Pickup 566
API, XML 959 Call Transfer 499
ATA(Cisco Analog Telephone Adapters) 147 Call Transfer Blocking 503
Audio Paging 785 Call Transfer Support 499
Authentication, Phone 387 Call Waiting 568
Auto-Answer, Headset 745 Call Waiting for Overlaid Ephone-dns 581
Automatic Agent Status Not-Ready, Ephone Hunt Callback Busy Subscriber 569
Groups 575 Called-Name Display 689
Automatic Line Selection 461 Caller ID Blocking 639
Auto-Registration Blocking 129 Call-Park Blocking 489
Call-Park Redirect 489
Call-Waiting Beep 568
B Call-Waiting Ring 569
Channel Huntstop 565
Backup Router 119 Cisco IP Communicator 147
BLF notification 797 Cisco VG 224 147
Blocking Call Transfer 503 Conference Gain Control 649
Blocking Caller ID 639 Conference Initiator Drop-Off Control 647
Blocking Call-Park 489 Conferencing 647
Blocking Calls Based on Date and Time 467 Configuration Files 120, 245
Blocking Features 467, 829 Customized Background Images 893
Blocking Local Directory 689 Customizing Feature Buttons 893
Blocking, Automatic Registration 129
Bulk-Loading Speed-Dial Numbers 858
Busy Timeout 119 D
F
J
FAC (feature access code) 729
fallback support for Cisco Unified Communica- Join Ephone Hunt Groups 575
tions Manager 943
Fax Relay 737
Feature Blocking 829 K
Feature Buttons,URL Provisioning 893
Keep-Conference Options 649
Feature Control 829
Key System 50, 176
Feature Ring 709
Keyswitch 50, 176
Files, Configuration 245
KPML 147
Fixed Line/Feature Button Set 893
Flash Soft Key 829
Forwarding 499
L P
N Q
Network Locales 287 QSIG Supplementary Services 499
Network Time Protocol 91, 119, 139
Night Service 579
R
S U
Secondary Dial Tone 267 URL Provisioning for Feature Buttons 893
Secondary Router 119 User-Defined Locales 287
Security 387
Selective Call Forwarding 500
Sequential Ephone Hunt Groups 571 V
Session Transport Protocol 147
Shared Lines 152 Vendor Configuration Parameters 895
Shared-line Overlay Ephone-dns 581 Video Support 867
Silent Ring 563 Voice Hunt Groups 569
SIP Dial Plans 147 Voice Mail Integration 355
SIP Trunks 91 Voice Translation Rules and Profiles 267
Soft Keys 829
Software-based Conferencing 647
Speed Dial 847 X
Abbreviated Dialing 856
Bulk Loading 858 XML Application Programming Interface 959
Local Speed Dial 852 XML Configuration Files 69
Monitor-Line Button 850
Personal Speed Dial 855
Speed-Dial Buttons 856
SRST Fallback Support Using Cisco Unified CME
943
SRTP 387
System Message Display 893
System-Defined Locales 287
This roadmap lists the features documented in the Cisco Unified Communications Manager Express
System Administrator Guide and maps them to the modules in which they appear.
Note Table 1 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
CCVP, the Cisco Logo, and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn is a
service mark of Cisco Systems, Inc.; and Access Registrar, Aironet, BPX, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, Cisco, the Cisco
Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity,
Enterprise/Solver, EtherChannel, EtherFast, EtherSwitch, Fast Step, Follow Me Browsing, FormShare, GigaDrive, GigaStack, HomeLink, Internet
Quotient, IOS, iPhone, IP/TV, iQ Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, LightStream, Linksys, MeetingPlace, MGX,
Networking Academy, Network Registrar, Packet, PIX, ProConnect, RateMUX, ScriptShare, SlideCast, SMARTnet, StackWise, The Fastest Way to
Increase Your Internet Quotient, and TransPath are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other
countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Cisco and any other company. (0612R)
Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and
figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and
coincidental.
Cisco Unified Communications Manager Express (formerly known as Cisco Unified CallManager
Express) is a call-processing application in Cisco IOS software that enables Cisco routers to deliver
key-system or hybrid PBX functionality for enterprise branch offices or small businesses.
Contents
• Information About Cisco Unified CME, page 47
• Where to Go Next, page 52
• Additional References, page 52
• Obtaining Documentation, Obtaining Support, and Security Guidelines, page 54
Figure 1 Cisco Unified CME for the Small- and Medium-Size Office
Telephone Telephone
Fax
PSTN
PCs
Gatekeeper
Figure 2 shows a branch office with several Cisco Unified IP phones connected to a
Cisco IAD2430 series router with Cisco Unified CME. The Cisco IAD2430 router is connected to a
multiservice router at a service provider office, which provides connection to the WAN and PSTN.
Telephone Telephone
IP
PSTN network
Fax
Voice
switch
Cisco IAD2430
Service
T1/DSL/Cable
provider
IAD V office
IP IP IP
Voice-mail
Gatekeeper server
146627
PCs
A Cisco Unified CME system uses the following basic building blocks:
• Ephone or voice register pool—A software concept that usually represents a physical telephone,
although it is also used to represent a port that connects to a voice-mail system, and provides the
ability to configure a physical phone using Cisco IOS software. Each phone can have multiple
extensions associated with it and a single extension can be assigned to multiple phones. Maximum
number of ephones and voice register pools supported in a Cisco Unified CME system is equal to
the maximum number of physical phones that can be connected to the system.
• Directory number—A software concept that represents the line that connects a voice channel to a
phone. A directory number represents a virtual voice port in the Cisco Unified CME system, so the
maximum number of directory numbers supported in Cisco Unified CME is the maximum number
of simultaneous call connections that can occur. This concept is different from the maximum number
of physical lines in a traditional telephony system.
Licenses
You must purchase a base Cisco Unified CME feature license and phone user licenses that entitle you to
use Cisco Unified CME.
Note To support H.323 call transfers and forwards to network devices that do not support the H.450 standard,
such as Cisco Unified Communications Manager, a tandem gateway is required in the network. The
tandem gateway must be running Cisco IOS release 12.3(7)T or a later release and requires the
Integrated Voice and Video Services feature license (FL-GK-NEW-xxx), which includes H.323
gatekeeper, IP-to-IP gateway, and H.450 tandem functionality.
PBX Model
The simplest case is the PBX model, in which most of the IP phones in your system have a single unique
extension number. Incoming PSTN calls are routed to a receptionist at an attendant console or to an
automated attendant. Phone users may be in separate offices or be geographically separated and
therefore often use the telephone to contact each other.
For this model, we recommend that you configure directory numbers as dual-lines so that each button
that appears on an IP phone can handle two concurrent calls. The phone user toggles between calls using
the blue navigation button on the phone. Dual-line directory numbers enable your configuration to
support call waiting, call transfer with consultation, and three-party conferencing (G.711 only).
Figure 3 shows a PSTN call that is received at the Cisco Unified CME router, which sends it to the
designated receptionist or automated attendant (1), which then routes it to the requested extension (2).
FXO ports
1
2
Cisco Unified CME Receptionist or
automated attendant
IP IP IP
146456
Extension Extension Extension
1001 1002 1003
For configuration information, see the “How to Configure Phones for a PBX System” section on
page 158.
Keyswitch Model
In a keyswitch type of system, you can set up most of your phones to have a nearly identical
configuration, in which each phone is able to answer any incoming PSTN call on any line. Phone users
are generally in close proximity and have little need to use the telephone to contact each other.
For example, a 3x3 keyswitch system has three PSTN lines shared across three telephones, such that all
three PSTN lines appear on each of the three telephones. This permits an incoming call on any PSTN
line to be directly answered by any telephone—without the aid of a receptionist, auto-attendant or the
use of (expensive) DID lines. Also, the lines act as shared lines—a call can be put on hold on one phone
and resumed on another phone without invoking call transfer.
In the keyswitch model, the same directory numbers are assigned to all IP phones. When an incoming
call arrives, it rings all available IP phones. When multiple calls are present within the system at the
same time, each individual call (ringing or waiting on hold) is visible and can be directly selected by
pressing the corresponding line button on an IP phone. In this model, calls can be moved between phones
simply by putting the call on hold at one phone and selecting the call using the line button on another
phone. In a keyswitch usage model, it is often not appropriate to use the dual-line option because the
PSTN lines to which the directory numbers correspond do not themselves support dual-line
configuration. Use of the dual-line option also makes configuration of call-coverage (hunting) behaviors
more complex.
You configure the keyswitch model by creating a set of directory numbers that correspond one-to-one
with your PSTN lines. Then you configure your PSTN ports to route incoming calls to those ephone-dns.
The maximum number of PSTN lines that you can assign in this model can be limited by the number of
available buttons on your IP phones. If so, the overlay option may be useful for extending the number of
lines that can be accessed by a phone.
Figure 4 shows an incoming call from the PSTN (1), which is routed to extension 1001 on all three
phones (2).
FXO ports
1
IP IP IP
Extension Extension Extension
1001 1001 1001
146457
For configuration information, see the “How to Configure Phones for a Key System” section on
page 176.
Hybrid Model
PBX and keyswitch configurations can be mixed on the same IP phone and can include both unique
per-phone extensions for PBX-style calling and shared lines for keyswitch-style call operations.
Single-line and dual-line directory numbers can be combined on the same phone.
In the simplest keyswitch deployments, individual telephones do not have private extension numbers.
Where key system telephones do have individual lines, the lines are sometimes referred to as intercoms
rather than as extensions. The term “Intercom” is derived from Internal Communication; there is no
assumption of the common ''intercom press-to-talk'' behavior of auto dial or auto answer in this context,
although those options may exist.
For key systems that have individual intercom (extension) lines, PSTN calls can usually be transferred
from one key system phone to another using the intercom (extension) line. When Call Transfer is
invoked in the context of a connected PSTN line, the outbound consultation call is usually placed from
the transferrer phone to the transfer-to phone using one of the phone's intercom (extension) line buttons.
When the transferred call is connected to the transfer-to phone and the transfer is committed (the
transferrer hangs up), the intercom lines on both phones are normally released and the transfer-to call
continues in the context of the original PSTN line button (all PSTN lines are directly available on all
phones). This behavior allows the transferred call to be put on hold (on the PSTN line button) and then
subsequently resumed from another phone that shares that PSTN line.
For example, you can design a 3x3 keyswitch system as shown in Figure 4 and then add another, unique
extension on each phone (Figure 5). This setup will allow each phone to have a “private” line to use to
call the other phones or to make outgoing calls.
FXO ports
1
IP IP IP
Extension Extension Extension
1001 1001 1001
1002 1002 1002
146458
1003 1003 1003
1004 1005 1006
Where to Go Next
Before configuring Cisco Unified CME, see “Before You Begin” on page 55.
Additional References
The following sections provide references related to Cisco Unified CME.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Cisco IOS voice troubleshooting • Cisco IOS Voice Troubleshooting and Monitoring Guide
Related Websites
Related Topic Title and Location
Cisco IOS configuration examples Cisco Systems Technologies website at
http://cisco.com/en/US/tech/index.html
Note From the website, select a technology category and
subsequent hierarchy of subcategories, then click Technical
Documentation > Configuration Examples.
MIBs
MIBs MIBs Link
CISCO-CCME-MIB To locate and download MIBs for selected platforms, Cisco IOS
releases, and feature sets, use Cisco MIB Locator found at the
MIB CISCO-VOICE-DIAL-CONTROL-MIB
following URL:
http://www.cisco.com/go/mibs
This module describes general decisions that you should make before you configure Cisco Unified
Communications Manager Express (Cisco Unified CME).
Contents
• Information About Planning Your Configuration, page 55
• How to Configure Cisco Unified CME, page 60
• Feature Summary, page 64
• Planning Worksheets, page 67
System Design
Traditional telephony systems are based on physical connections and are therefore limited in the types
of phone services that they can offer. Because phone configurations and directory numbers in a
Cisco Unified CME system are software entities and because the audio stream is packet-based, an almost
limitless number of combinations of phone numbers, lines, and phones can be planned and implemented.
Cisco Unified CME systems can be designed in many ways. The key is to determine the total number of
simultaneous calls you want to handle at your site and at each phone at your site, and how many different
directory numbers and phones you want to have. Even a Cisco Unified CME system has its limits,
however. Consider the following factors in your system design:
• Maximum number of phones—This number corresponds to the maximum number of devices that
can be attached. The maximum is platform- and version-dependent. To find the maximum for your
platform and version, see the appropriate Cisco CME Supported Firmware, Platforms, Memory, and
Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap0918
6a0080189132.html.
• Maximum number of directory numbers—This number corresponds to the maximum number of
simultaneous call connections that can occur. The maximum is platform- and version-dependent. To
find the maximum for your platform and version, see the appropriate Cisco CME Supported
Firmware, Platforms, Memory, and Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap0918
6a0080189132.html.
• Telephone number scheme—Your numbering plan may restrict the range of telephone numbers or
extension numbers that you can use. For example, if you have DID, the PSTN may assign you a
certain series of numbers.
• Maximum number of buttons per phone—You may be limited by the number of buttons and phones
that your site can use. For example, you may have two people with six-button phones to answer 20
different telephone numbers.
The flexibility of a Cisco Unified CME system is due largely to the different types of directory numbers
(DNs) that you can assign to phones in your system. By understanding types of DNs and considering
how they can be combined, you can create the complete call coverage that your business requires. For
more information about DNs, see“Configuring Phones to Make Basic Calls” on page 147.
After setting up the DNs and phones that you need, you add optional Cisco Unified CME features to
create a telephony environment that enhances your business objectives. Cisco Unified CME systems are
able to integrate with the PSTN and with your business requirements to allow you to continue using your
existing number plans, dialing schemes, and call coverage patterns.
When creating number plans, dialing schemes, and call coverage patterns in Cisco Unified CME, there
are several factors that you must consider:
• Is there an existing PBX or Key System that you are replacing and want to emulate?
• Number of phones and phone users to be supported?
• Do you want to use single-line or dual-line DNs?
• What protocols does your voice network support?
• Which call transfer and forwarding methods must be supported?
• What existing or preferred billing method do you want to use for transferred and forwarded calls?
• Do you need to optimize network bandwidth or minimize voice delay?
Because these factors can limit your choices for some of the configuration decisions that you will make
when you create of a dialing plan, see the Cisco Unified CallManager Express Solution Reference
Network Design Guide to help you understand the effect these factors have on your Cisco Unified CME
implementation.
.
Table 2 Comparison of Configuration Methods for Cisco Unified CME
Table 3 Parameters and Features Supported by Cisco Unified Communications Express - QCT
– TFTP address
The GUI supports authentication, authorization, and accounting (AAA) authentication for system
administrators through a remote server capability. If authentication through the server fails, the local
router is searched.
Cisco Unified CME GUI must be installed and set up before it can be used. Instructions for using the
Cisco Unified GUI are in online help for the GUI.
For information about using the Cisco Unified CME GUI, see the “Using Cisco Unified CME GUI to
Modify or Maintain Configuration” section on page 63.
Note To support H.323 call transfers and forwards to network devices that do not support the H.450 standard,
such as Cisco Unified Communications Manager, a tandem gateway is required in the network. The
tandem gateway must be running Cisco IOS release 12.3(7)T or a later release and requires the
Integrated Voice and Video Services feature license (FL-GK-NEW-xxx), which includes H.323
gatekeeper, IP-to-IP gateway, and H.450 tandem functionality.
Prerequisites
• Hardware and software to establish a physical or virtual console connection to the Cisco router using
a terminal or PC running terminal emulation is available and operational.
• To establish a physical console connection, attach a terminal or PC running terminal emulation to
the console port of the router. For more information on cabling, and details about how to connect a
terminal to the console port or the AUX port, see Cabling Guide for Console and Aux Ports on Cisco
Routers.
For connecting to the router to be configured, use the following terminal settings:
– 9600 baud rate
– No parity
– 8 data bits
– 1 stop bit
– No flow control
• We recommend that you complete the worksheets to gather required site-specific information for
the Cisco router to be configured. See the worksheet set at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_configuration_guide09186a00807
59201.html#wp1007671.
Note Even though the worksheet set is for configuring a Typical (recommended) installation of an IP
telephony system using Cisco Unified Communications Express - QCT, the information is required to
create an initial configuration using any of the configuration methods.
Prerequisites
• Your IP network is operational and you can access Cisco web.
• A PC with Microsoft® Internet Explorer 5.5 or later is connected, using a serial cable, to the console
port of Cisco router to be configured. If you need assistance in connecting your PC to your router's
console port, refer to the Install and Upgrade Guide for the Cisco router.
• The Block Pop-up Windows feature for Microsoft® Internet Explorer must be disabled.
• You must be a member of Administrators group under User Account settings for your PC.
• You must have a valid Cisco CCO account.
• The factory-default configuration is loaded in nonvolatile memory and in the running configuration
of the Cisco router to be configured.
Note Cisco Unified Communications Express - QCT can be used to restore the factory default configuration
on router to be configured.
• If you are using Cisco Unified Communications Express - QCT to upload firmware files for
Cisco Unified IP phones after uploading the generated configuration, all Cisco firmware files to be
uploaded must be installed in the folder named Phoneloads, within the local folder in which
Cisco Unified Communications Express - QCT is installed.
• Worksheets for Cisco Unified Communications Express - QCT are complete with required
site-specific information for the Cisco router to be configured.
Restrictions
• Cisco Unified Communications Express - QCT cannot be used to perform routine additions and
changes associated with employee turnover.
• Cisco Unified Communications Express - QCT can configure only a subset of features of
Cisco Unified CME. You must use Cisco IOS commands to add features or to modify the
configuration.
What to Do Next
After using Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip the following modules in the Cisco Unified CME System Administrator Guide when you
use Cisco IOS commands to modify the configuration:
• Defining Network Parameters
• Configuring System-Level Parameters
• Configuring Phones to Make Basic Calls
• Creating Phone Configurations Using Extension Assigner
• Configuring Dialing Plans
• Adding Features, but only for the following:
– Configuring Call Blocking
– Configuring Call Park
– Configuring Call Transfer and Forwarding
– Configuring Call-Coverage Features
– Configuring Caller ID Blocking
– Configuring Conferencing
– Configuring Intercom Lines
– Configuring Music on Hold
– Configuring Paging
• Enabling the GUI
• Configuring Voice-Mail Support
Prerequisites
• Cisco CME 3.2 or a later version.
• Files required for the operation of the GUI must be copied into flash memory on the router. For
information about files, see “Installing and Upgrading Cisco Unified CME Software” on page 69.
Restrictions
• The web browser that you use to access the GUI must be Microsoft Internet Explorer 5.5 or a later
version. No other type of browser can be used to access the GUI.
• Cannot provision voice features such as digit translation, call routing, and class of restriction.
• Cannot provision data features such as DHCP, IP addressing, and VLANs.
• Can only provision IP phones that are registered to Cisco Unified CME. Cannot use bulk
administration to import multiple phones at the same time. Cannot manage IP phone firmware.
• Requires manual upgrade of files in flash memory of router if Cisco Unified CME is upgraded to
later version.
• Other minor limitations, such as:
– If you use an XML configuration file to create a customer administrator login, the size of that
XML file must be 4000 bytes or smaller.
– The password of the system administrator cannot be changed through the GUI. Only the
password of a customer administrator or a phone user can be changed through the GUI.
– If more than 100 phones are configured, choosing to display all phones will result in a long
delay before results are shown.
Feature Summary
Table 4 contains a list of commonly configured features in Cisco Unified CME and the module in which
they appear in this guide. For a detailed list of features, with links to corresponding information in this
guide, see “Cisco Unified CME Features Roadmap” on page 37.
Planning Worksheets
Before configuring Cisco Unified CME, we recommend that you complete the worksheets to gather
required site-specific information for the Cisco router to be configured. See the worksheet set at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_configuration_guide09186a00807592
01.html#wp1007671.
Note Even though the worksheet set is for configuring a Typical (recommended) installation of an IP
telephony system using Cisco Unified Communications Express - QCT, the information is required to
create an initial configuration using any of the configuration methods.
This chapter explains how to install Cisco Unified Communications Manager Express
(Cisco Unified CME) software and how to upgrade phone firmware for Cisco Unified IP phones.
Contents
• Prerequisites for Installing Cisco Unified CME Software, page 69
• Information About Cisco Unified CME Software, page 70
• How to Install and Upgrade Cisco Unified CME Software, page 74
• Additional References, page 90
Basic Files
A tar archive contains the basic files you need for Cisco Unified CME. Be sure to download the correct
version for the Cisco IOS software release that is running on your router. The basic tar archive generally
also contains the phone firmware files that you require, although you may occasionally need to download
individual phone firmware files. For information about installing Cisco Unified CME, see the
“Installing Cisco Unified CME Software” section on page 74.
GUI Files
A tar archive contains the files that you need to use the Cisco Unified CME graphical user interface
(GUI), which provides a mouse-driven interface for provisioning phones after basic installation is
complete. For installation information, see the “Installing Cisco Unified CME Software” section on
page 74.
Note Cisco Unified CME GUI files are version-specific; GUI files for one version of Cisco Unified CME are
not compatible with any other version of Cisco Unified CME. When downgrading or upgrading
Cisco Unified CME, the GUI files for the old version must be overwritten with GUI files that match the
Cisco Unified CME version that is being installed.
New IP phones are shipped from Cisco with a default manufacturing SCCP image. When a IP phone
downloads its configuration profile, the phone compares the phone firmware mentioned in the
configuration profile with the firmware already installed on the phone. If the firmware version differs
from the one that is currently loaded on the phone, the phone contacts the TFTP server to upgrade to the
new phone firmware and downloads the new firmware before registering with Cisco Unified CME.
Generally, phone firmware files are included in the Cisco Unified CME software archive that you
download. They can also be posted on the software download website as individual files or archives.
Early versions of Cisco phone firmware for SCCP and SIP IP phones had filenames as follows:
• SCCP firmware—P003xxyy.bin
• SIP firmware—P0S3xxyy.bin
In both bases, x represents the major version, and y represented the minor version. The third character
represents the protocol, “0” for SCCP or “S” for SIP.
In later versions, the following conventions are used:
• SCCP firmware—P003xxyyzzww, where x represents the major version, y represents the major
subversion, z represents the maintenance version, and w represents the maintenance subversion.
• SIP firmware—P0S3-xx-y-zz, where x represents the major version, y represents the minor version,
and z represents the subversions.
• The third character in a filename—Represents the protocol, “0” for SCCP or “S” for SIP.
There are exceptions to the general guidelines. For Cisco ATA, the filename begins with AT. For
Cisco Unified IP Phone 7002, 7905, and 7912, the filename can begin with CP.
Signed and unsigned versions of phone firmware are available for certain phone types. Signed binary
files support image authentication, which increases system security. We recommend signed versions if
your version of Cisco Unified CME supports them. Signed binary files have .sbn file extensions, and
unsigned files have .bin file extensions.
For Java-based IP phones, such as the Cisco Unified IP Phone 7911, 7941, 7941GE, 7961, 796GE, 7970,
and 7971, the firmware consists of multiple files including JAR and tone files. All of the firmware files
for each phone type must be downloaded the TFTP server before they can be downloaded to the phone.
The following example shows a list of phone firmware files that are installed in flash memory for the
Cisco Unified IP Phone 7911:
tftp server-flash:SCCP11.7-2-1-0S.loads
tftp server-flash:term06.default.loads
tftp server-flash:term11.default.loads
tftp server-flash:cvm11.7-2-0-66.sbn
tftp server-flash:jar11.7-2-0-66.sbn
tftp server-flash:dsp11.1-0-0-73.sbn
tftp server-flash:apps11.1-0-0-72.sbn
tftp server-flash:cnu11.3-0-0-81.sbn
However, you only specify the filename for the image file when configuring Cisco Unified CME. For
Java-based IP phones, the following naming conventions are used for image files:
• SCCP firmware—TERMnn.xx-y-z-ww or SCCPnn.xx-y-zz-ww, where n represents the phone type,
x represents the major version, y represents the major subversion, z represents the maintenance
version, and w represents the maintenance subversion.
The following example shows how to configure Cisco Unified CME so that the Cisco Unified IP Phone
7911 can download the appropriate SCCP firmware from flash memory:
Router(config)# telephony-service
Router(config-telephony)#load 7911 SCCP11.7-2-1-0S
The phone firmware filenames for each phone type and Cisco Unified CME version are listed in the
appropriate Cisco CME Supported Firmware, Platforms, Memory, and Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
For information about installing firmware files, see the “Installing Cisco Unified CME Software”
section on page 74.
For information about configuring Cisco Unified CME for upgrading between versions or converting
between SCCP and SIP, see the “How to Install and Upgrade Cisco Unified CME Software” section on
page 74.
IXML Template
The file called xml.template can be copied and modified to allow or restrict specific GUI functions to
customer administrators, a class of administrative users with limited capabilities in a
Cisco Unified CME system. This file is included in both tar archives (cme-basic-... and cme-gui-...). To
install the file, see the “Installing Cisco Unified CME Software” section on page 74.
Script Files
Archives containing Tcl script files are listed individually on the Cisco Unified CME software download
website. For example, the file named app-h450-transfer.2.0.0.9.zip.tar contains a script that adds H.450
transfer and forwarding support for analog FXS ports.
The Cisco Unified CME Basic Automatic Call Distribution and Auto Attendant Service (B-ACD)
requires a number of script files and audio files, which are contained in a tar archive with the name
cme-b-acd-.... For a list of files in the archive and for more information about the files, see the
addition, Cisco Unified Communications Express - QCT recognizes any Advanced Integrated Module
(AIM) or NM-CUE module installed in the router, thus providing voice-mail and Auto Attendant (AA)
capability to the Cisco Unified CME system.
After all the necessary information is entered, Cisco Unified Communications Express - QCT generates
all of the required configuration commands which you can upload to the Cisco router to be configured
or save as a template file to use to configure additional systems with similar system parameters.
For information about installing and using Cisco Unified Communications Express - QCT, see the
Cisco Unified Communications Express - QCT User Guide.
Note Customers who purchase a router bundle enabled with Cisco Unified CME will have the necessary
Cisco Unified CME files installed at time of manufacture.
SUMMARY STEPS
DETAILED STEPS
Step 1 Go to http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key.
Step 2 Select the file to download.
Step 3 Download zip file to tftp server.
Step 4 Use the zip program to extract the file to be installed, then:
a. If the file is an individual file, use the copy command to copy the files to router flash:
Router# copy tftp://x.x.x.x/P00307020300.sbn flash:
b. If the file is a tar file, use the archive tar command to extract the files to flash memory.
Router# archive tar /xtract source-url flash:/file-url
Step 5 Verify the installation. Use the show flash: command to list the files installed in in flash memory.
Router# show flash:
What to Do Next
• If you installed Cisco Unified CME software and Cisco Unified CME is not configured on your
router, see “Defining Network Parameters” on page 91.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SCCP protocol
to receive and place calls and the firmware version must be upgraded to a recommended version, or
if the phones to be connected to Cisco Unified CME are brand new, out-of-the-box, the phone
firmware preloaded at the factory must be upgraded to the recommended version before your phones
can complete registration, see the “SCCP: Upgrading or Downgrading Phone Firmware Between
Versions” section on page 76.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SIP protocol to
receive and place calls and the firmware version must be upgraded to a recommended version, see
the “SIP: Upgrading or Downgrading Phone Firmware Between Versions” section on page 77.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SCCP protocol
to receive and place calls and you now want some or all of these phones to use the SIP protocol, the
phone firmware for each phone type must be upgraded from SCCP to the recommended SIP version
before the phones can register. See the “SCCP: Converting Phone Firmware to SIP” section on
page 81.
• If Cisco Unified IP phones to be connected to Cisco Unified CME are using the SIP protocol and
are brand new, out-of-the-box, the phone firmware preloaded at the factory must be upgraded to the
recommended SIP version before your SIP phones can complete registration. See the “SCCP:
Converting Phone Firmware to SIP” section on page 81.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SIP protocol to
receive and place calls and you now want some or all of these phones to use the SCCP protocol, the
phone firmware for each phone type must be upgraded from SIP to the recommended SCCP version
before the phones can register. See the “SIP: Converting Phone to SCCP” section on page 84.
Prerequisites
Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of the
TFTP server from which the phones download their configuration profiles. For information about
installing firmware files in flash memory, see the “Installing Cisco Unified CME Software” section on
page 74.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. load phone-type firmware-file
5. create cnf
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 tftp-server flash: file-name Enables TFTP file sharing for new phone firmware files.
• A separate tftp-server flash command is required for
Example: each firmware file to be downloaded to this phone.
Router(config)# tftp-server
flash:P00307020300.loads Router(config)#
tftp-server flash:P00307020300.sb2
Router(config)# tftp-server
flash:P00307020300.sbn
Router(config)# tftp-server
flash:P00307020300.bin
Step 4 telephony service Enters telephone-service configuration mode.
Example:
Router(config)# telephony service
Step 6 create cnf-files Builds XML configuration files required for SCCP phones.
Example:
Router(config-telephony)# create cnf-files
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end
What to Do Next
• If the Cisco Unified IP phone to be upgraded is not configured in Cisco Unified CME, see “How to
Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phone is already configured in Cisco Unified CME and can make and receive
calls, you are ready to reboot the Cisco Unified IP phones to download the phone firmware to the
phone. See “Resetting and Restarting Phones” on page 257.
Prerequisites
Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of the
TFTP server from which the phones will download their configuration profiles. For information about
installing firmware files in flash memory, see the “Installing Cisco Unified CME Software” section on
page 74.
Restrictions
• Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco ATA—Signed load starts
from SIP v1.1. After you upgrade the firmware to a signed load, you cannot downgrade the firmware
to an unsigned load.
• Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G—Signed load starts from SIP
v5.x. Once you upgrade the firmware to a signed load, you cannot downgrade the firmware to an
unsigned load.
• The procedures for upgrading phone firmware files for SIP phones is the same for all
Cisco Unified IP phones. For other limits on firmware upgrade between versions, see the phone
firmware upgrade matrix at:
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/prod_installation_guides_list.html.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. mode cme
5. load phone-type firmware-file
6. upgrade
7. Repeat Steps 5 and 6.
8. file text
9. create profile
10. exit
11. voice register pool tag
12. reset
13. exit
14. voice register global
15. no upgrade
16. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-register-global)# no upgrade
Step 16 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
Examples
The following example shows the configuration steps for upgrading firmware for a Cisco Unified IP
Phone 7960G or Cisco Unified IP Phone 7940G from SIP 5.3 to SIP 6.0, then from SIP 6.0 to SIP 7.4:
Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-06-0-00
Router(config-register-global)# upgrade
Router(config-register-global)# load 7960 P0S3-07-4-00
Router(config-register-global)# create profile
The following example shows the configuration steps for downgrading firmware for a Cisco Unified IP
Phone 7960/40 from SIP 7.4 to SIP 6.0:
Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-06-0-00
Router(config-register-global)# upgrade
Router(config-register-global)# create profile
What to Do Next
• If the Cisco Unified IP phone to be upgraded is not configured in Cisco Unified CME, see “How to
Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phone is already configured in Cisco Unified CME and can make and receive
calls, you are ready to reboot the Cisco Unified IP phones to download the phone firmware to the
phone. See “Resetting and Restarting Phones” on page 257.
Note If codec values for the dial peers of a connection do not match, the call fails. The default codec for the
POTS dial peer for an SCCP phone is G.711 and the default codec for a VoIP dial peer for a SIP phone
is G.729. If neither the SCCP phone nor the SIP phone in Cisco Unified CME has been specifically
configured to change the codec, calls between the two IP phones on the same router will produce a busy
signal caused by the mismatched default codecs. To avoid codec mismatch, specify the codec for IP
phones in Cisco Unified CME. For configuration information, see the “Configuring Codec for Local
Calling Between SIP and SCCP Phones” section on page 174.
Prerequisites
• Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of
the TFTP server from which the phones download their configuration profiles. For information
about installing firmware files in flash memory, see the “Installing Cisco Unified CME Software”
section on page 74.
• Cisco Unified IP Phone 7940Gs and Cisco Unified IP Phone 7960Gs—If these IP phones are
already configured in Cisco Unified CME to use the SCCP protocol, the SCCP phone firmware on
the phone must be version 5.x. If required, upgrade the SCCP phone firmware to 5.x before
upgrading to SIP.
SUMMARY STEPS
1. enable
2. configure terminal
3. no ephone ephone-tag
4. exit
5. no ephone-dn dn-tag
6. exit
7. voice register global
8. mode cme
9. load phone-type firmware-file
10. upgrade
11. Repeat previous two steps.
12. create profile
DETAILED STEPS
Example:
Router# configure terminal
Step 3 no ephone ephone-tag (Optional) Disables the ephone and removes the ephone
configuration.
Example: • Required only if the Cisco Unified IP phone to be
Router (config)# no ephone 23 configured is already connected to Cisco Unified CME
and is using SCCP protocol.
• ephone-tag—Particular IP phone to which this
configuration change will apply.
Step 4 exit (Optional) Exits from the current command mode to the
next highest mode in the configuration mode hierarchy.
Example: • Required only if you performed the previous step.
Router(config-ephone)# exit
Step 5 no ephone-dn dn-tag (Optional) Disables the ephone-dn and removes the
ephone-dn configuration.
• Required only if this directory number is not now nor
will be associated to any SCCP phone line, intercom
line, paging line, voice-mail port, or message-waiting
indicator (MWI) connected to Cisco Unified CME.
• dn-tag—Particular configuration to which this change
will apply.
Step 6 exit (Optional) Exits from the current command mode to the
next highest mode in the configuration mode hierarchy.
Example: • Required only if you performed the previous step.
Router(config-ephone-dn)# exit
Step 7 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 8 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 12 create profile Generates provisioning files required for SIP phones and
writes the file to the location specified with the tftp-path
command.
Example:
Router(config-register-global;)# create profile
Step 13 file text (Optional) Generates ASCII text files for Cisco Unified IP
Phones 7905 and 7905G, Cisco Unified IP Phone 7912 and
Cisco Unified IP Phone 7912G, Cisco ATA-186, or
Example:
Router(config-register-global)# file text
Cisco ATA-188.
• Default—System generates binary files to save disk
space.
Step 14 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
Examples
The following example shows the configuration steps for converting firmware on an Cisco Unified IP
phone already connected in Cisco Unified CME and using the SCCP protocol, from SCCP 5.x to SIP 7.4:
Router(config)# telephony-service
Router(config-telephony)# no create cnf
CNF files deleted
Router(config-telephony)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-07-4-00
Router(config-register-global)# upgrade
Router(config-register-global)# create profile
What to Do Next
After you configure the upgrade command, refer to the following statements to determine which task
to perform next.
• If the Cisco Unified IP phone to be upgraded is already connected in Cisco Unified CME and you
removed the SCCP configuration file for the phone but have not configured this phone for SIP in
Cisco Unified CME, see “How to Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phones to be upgraded are already configured in Cisco Unified CME, see
“Resetting and Restarting Phones” on page 257.
Note If codec values for the dial peers of a connection do not match, the call fails. The default codec for the
POTS dial peer for an SCCP phone is G.711 and the default codec for a VoIP dial peer for a SIP phone
is G.729. If neither the SCCP phone nor the SIP phone in Cisco Unified CME has been specifically
configured to change the codec, calls between the two IP phones on the same router will produce a busy
signal caused by the mismatched default codecs. To avoid codec mismatch, specify the codec for SIP and
SCCP phones in Cisco Unified CME. For more information, see “How to Configure Phones for a PBX
System” on page 158.
Prerequisites
• Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of
the TFTP server from which the phones will download their configuration profiles. For information
about installing firmware files in flash memory, see the “Installing Cisco Unified CME Software”
section on page 74.
• Cisco Unified IP Phone 7940Gs and Cisco Unified IP Phone 7960Gs—If these IP phones are
already configured in Cisco Unified CME to use the SIP protocol, the SIP phone firmware must be
version 7.x. See the “SIP: Upgrading or Downgrading Phone Firmware Between Versions” section
on page 77.
SUMMARY STEPS
1. enable
2. configure terminal
3. no voice register pool pool-tag
4. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 no voice register pool pool-tag Disables voice register pool and removes the voice pool
configuration.
Example: • pool-tag—Unique sequence number for a particular
Router(config)# no voice register pool 1 SIP phone to which this configuration change will
apply.
Step 4 end Exits from the current command mode to the next highest
mode in the configuration mode hierarchy.
Example:
Router(config-register-pool)# end
Generating an SCCP XML Configuration File for Upgrading from SIP to SCCP
To create an ephone entry and generate a new SCCP XML configuration file for upgrading a particular
Cisco Unified IP phone in Cisco Unified CME from SIP to SCCP, perform the steps in this task.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. exit
5. tftp-server flash firmware-file
6. telephony service
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone dn 1 ephone-dn during configuration tasks. The maximum
number of ephone-dns in Cisco Unified CME is version
and platform specific. Type ? to display range.
Step 4 exit Exits from the current command mode to the next highest
mode in the configuration mode hierarchy.
Example:
Router(config-ephone-dn)# exit
Step 5 tftp-server flash: file-name Enables TFTP file sharing for new phone firmware files.
• A separate tftp-server flash command is required for
Example: each firmware file to be downloaded to this phone.
Router(config)# tftp-server
flash:P00307020300.loads
Router(config)# tftp-server
flash:P00307020300.sb2
Router(config)# tftp-server
flash:P00307020300.sbn
Router(config)# tftp-server
flash:P00307020300.bin
Step 6 telephony service Enters telephone-service configuration mode.
Example:
Router(config)# telephony service
Step 7 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each IP phone
Example: type.
Router(config-telephony)# load 7960-7940
P00307020300
• firmware-file—Filename to be associated with the
specified IP phone type.
• Do not use the .sbin or .loads file extension except for
Cisco ATA and Cisco Unified IP Phone 7905 and 7912
Example:
Router(config-telephony)# create cnf-files
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end
Examples
The following example shows the configuration steps for upgrading firmware for a Cisco Unified IP
Phone 7960G from SIP to SCCP. First the SIP firmware is upgraded to SIP 6.3 and from SIP 6.3 to SIP
7.4; then, the phone firmware is upgraded from SIP 7.4 to SCCP 7.2(3). The SIP configuration profile is
deleted and a new ephone configuration profile is created for the Cisco Unified IP phone.
Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-06-0-00
Router(config-register-global)# upgrade
Router(config-register-global)# load 7960 P0S3-07-4-00
Router(config-register-global)# exit
Router(config)# no voice register pool 1
Router(config-register-pool)# exit
Router(config)# voice register global
Router(config-register-global)# no upgrade
Router(config-register-global)# exit
Router(config)# ephone-dn 1
Router(config-ephone-dn)# exit
Router(config)# tftp-server flash:P00307020300.loads
Router(config)# tftp-server flash:P00307020300.sb2
Router(config)# tftp-server flash:P00307020300.sbn
Router(config)# tftp-server flash:P00307020300.bin
Router(config)# telephony service
Router(config-telephony)# load 7960-7940 P00307000100
Router(config-telephony)# create cnf-files
What to Do Next
After you configure the upgrade command:
• If the Cisco Unified IP phone to be upgraded is already connected in Cisco Unified CME and you
removed the SIP configuration file for the phone and have not configured the SCCP phone in
Cisco Unified CME, see “How to Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phones to be upgraded are already configured in Cisco Unified CME, see
“Resetting and Restarting Phones” on page 257.
SUMMARY STEPS
1. show flash:
2. show ephone phone-load
DETAILED STEPS
Troubleshooting Tips
Use the debug tftp event command to troubleshoot an attempt to upgrade or convert Cisco phone
firmware files for SIP phones. The following sample from the debug tftp event command shows how
the Cisco phone firmware for a Cisco Unified IP Phone 7940G is upgraded from SCCP 5.X to SIP 6.3.
The configuration profiles are downloaded when a phone is rebooted or reset.
Router# debug tftp event
…
Router(config)#telephony-service
Router(config-telephony)#no create cnf
CNF files deleted
Router(config-telephony)#voice register global
Router(config-register-global)#load 7960 P0S3-06-3-00
Router(config-register-global)#upgrade
Router(config-register-global)#create profile
Router(config-register-global)#
*May 6 17:37:03.737: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP000ED7DF7932 IP:1.5.49.84
Socket:4
DeviceType:Phone has unregistered normally.
*May 6 17:37:35.949: TFTP: Looking for OS79XX.TXT
*May 6 17:37:36.413: TFTP: Opened system:/cme/sipphone/OS79XX.TXT, fd 4, size 13 for
process 81
The following sample from the debug tftp event command shows how the Cisco phone firmware for a
Cisco Unified IP Phone 7940G is upgraded from SIP 6.3 to SIP 7.0 after the phone is rebooted or reset:
Router# debug tftp event
…
Router(config-register-global)#load 7960 P003-07-4-00
Router(config-register-global)#upgrade
Router(config-register-global)#load 7960 P0S3-07-4-00
Router(config-register-global)#create profile
Router(config-register-global)#end
Router-2012#
*May 6 17:42:35.581: TFTP: Looking for OS79XX.TXT
*May 6 17:42:35.585: TFTP: Opened system:/cme/sipphone/OS79XX.TXT, fd 5, size 13 for
process 81
*May 6 17:42:35.585: TFTP: Finished system:/cme/sipphone/OS79XX.TXT, time 00:00:00 for
process 81
*May 6 17:42:35.969: TFTP: Looking for P003-07-4-00.sbn
*May 6 17:42:35.977: TFTP: Opened slot0:P003-07-4-00.sbn, fd 5, size 129876 for process 81
*May 6 17:42:37.937: TFTP: Finished slot0:P003-07-4-00.sbn, time 00:00:01 for process 81
*May 6 17:44:31.037: TFTP: Looking for CTLSEP000ED7DF7932.tlv
*May 6 17:44:31.057: TFTP: Looking for SEP000ED7DF7932.cnf.xml
*May 6 17:44:31.089: TFTP: Looking for SIP000ED7DF7932.cnf
*May 6 17:44:31.089: TFTP: Opened system:/cme/sipphone/SIP000ED7DF7932.cnf, fd 5, size 789
for process 81
*May 6 17:44:31.089: TFTP: Finished system:/cme/sipphone/SIP000ED7DF7932.cnf, time
00:00:00 for process 81
*May 6 17:44:31.125: TFTP: Looking for P0S3-07-4-00.loads
*May 6 17:44:31.133: TFTP: Opened slot0:P0S3-07-4-00.loads, fd 5, size 461 for process 81
*May 6 17:44:31.141: TFTP: Finished slot0:P0S3-07-4-00.loads, time 00:00:00 for process 81
*May 6 17:44:31.673: TFTP: Looking for P0S3-07-4-00.sb2
*May 6 17:44:31.681: TFTP: Opened slot0:P0S3-07-4-00.sb2, fd 5, size 592626 for process 81
*May 6 17:44:33.989: TFTP: Finished slot0:P0S3-07-4-00.sb2, time 00:00:02 for process 81
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
• Cisco Unified Communications Express - QCT User Guide
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
This chapter describes how to define parameters that enable Cisco Unified Communications Manager
Express (Cisco Unified CME) to work with your network.
Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip this module unless you want to modify the configuration to relay DHCP requests from IP
phones to a DHCP server on a different router.
Contents
• Prerequisites for Defining Network Parameters, page 91
• Information About Defining Network Parameters, page 92
• How to Define Network Parameters, page 95
• Configuration Examples for Network Parameters, page 114
• Where to Go Next, page 115
• Additional References, page 115
• Feature Information for Network Parameters, page 117
DHCP Service
When a Cisco Unified IP phone is connected to the Cisco Unified CME system, it automatically queries
for a Dynamic Host Configuration Protocol (DHCP) server. The DHCP server responds by assigning an
IP address to the Cisco Unified IP phone and providing the IP address of the TFTP server through DHCP
option 150. Then the phone registers with the Cisco Unified CME server and attempts to get
configuration and phone firmware files from the TFTP server.
For configuration information, perform only one of the following procedures to set up DHCP service for
your IP phones:
• If your Cisco Unified CME router is the DHCP server and you can use a single shared address pool
for all your DHCP clients, see the “Defining a Single DHCP IP Address Pool” section on page 98.
• If your Cisco Unified CME router is the DHCP server and you need separate pools for non-IP-phone
DHCP clients, see the “Defining a Separate DHCP IP Address Pool for Each DHCP Client” section
on page 100.
• If the Cisco Unified CME router is not the DHCP server and you want to relay DHCP requests from
IP phones to a DHCP server on a different router, see the “Defining a DHCP Relay” section on
page 102.
DTMF Relay
IP phones connected to Cisco Unified CME systems require the use of out-of-band DTMF relay to
transport DTMF (keypad) digits across VoIP connections. The reason for this is that the codecs used for
in-band transport may distort DTMF tones and make them unrecognizable. DTMF relay solves the
problem of DTMF tone distortion by transporting DTMF tones out-of-band, or separate, from the
encoded voice stream.
For IP phones on H.323 networks, DTMF is relayed using the H.245 alphanumeric method, which is
defined by the ITU H.245 standard. This method separates DTMF digits from the voice stream and sends
them as ASCII characters in H.245 user input indication messages through the H.245 signaling channel
instead of the RTP channel. For information about configuring a DTMF relay in a multisite installation,
see the “Configuring DTMF Relay for H.323 Networks in Multisite Installations” section on page 105.
To use remote voice-mail or IVR applications on SIP networks from Cisco Unified CME phones, the
DTMF digits used by the Cisco Unified CME phones must be converted to the RFC 2833 in-band DTMF
relay mechanism used by SIP phones. The SIP DTMF relay method is needed in the following situations:
• When SIP is used to connect a Cisco Unified CME system to a remote SIP-based IVR or voice-mail
application.
• When SIP is used to connect a Cisco Unified CME system to a remote SIP-PSTN voice gateway that
goes through the PSTN to a voice-mail or IVR application.
The requirement for out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones
natively support in-band DTMF relay as specified in RFC 2833.
To use voice mail on a SIP network that connects to a Cisco Unity Express system, which uses a
nonstandard SIP Notify format, the DTMF digits used by the Cisco Unified CME phones must be
converted to the Notify format. Additional configuration may be required for backward compatibility
with Cisco CME 3.0 and 3.1. For configuration information about enabling DTMF relay for SIP
networks, see “Configuring SIP Trunk Support” section on page 106.
When registering E.164 numbers in dial peers with an external registrar, you can also register them with
a secondary SIP proxy or registrar to provide redundancy. The secondary registration can be used if the
primary registrar fails. For configuration information, see the “Basic SIP Configuration” chapter in the
Cisco IOS SIP Configuration Guide.
Note No commands allow registration between the H.323 and SIP protocols.
By default, SIP gateways do not generate SIP Register messages, so the gateway must be configured to
register the gateway’s E.164 telephone numbers with an external SIP registrar. For information about
configuring the SIP gateway to register phone numbers with Cisco Unified CME, see the “Configuring
SIP Trunk Support” section on page 106.
Out-of-Dialog REFER
Out-of-dialog REFER (OOD-R) allows remote applications to establish calls by sending a REFER
message to Cisco Unified CME without an initial INVITE. After the REFER is sent, the remainder of
the call setup is independent of the application and the media stream does not flow through the
application. The application using OOD-R triggers a call setup request that specifies the Referee address
in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to
communicate with Cisco Unified CME is independent of the end-user device protocol which can be SIP,
SCCP, H.323, or POTS. Click-to-dial is an example of an application that can be created using OOD-R.
A click-to-dial application allows users to combine multiple steps into one click for a call setup. For
example, a user can click a web-based directory application from their PC to look up a telephone
number, off-hook their desktop phone, and dial the called number. The application initiates the call setup
without the user having to out-dial from their own phone. The directory application sends a REFER
message to Cisco Unified CME which sets up the call between both parties based on this REFER.
Figure 6 shows an example of OOD-R being used by a click-to-dial application. In this scenario, the
following events occur (refer to the event numbers in the illustration):
1. Remote user clicks to dial.
2. Application sends out-of-dialog REFER to Cisco Unified CME 1.
3. Cisco Unified CME 1 connects to SIP phone 1 (Referee).
4. Cisco Unified CME 1 sends INVITE to Cisco Unified CME 2.
5. Cisco Unified CME 2 sends INVITE to SIP phone 2 (Refer-Target) and the call is accepted.
6. Voice path is created between the two SIP phones.
1 IP
6 5
2
IP 3 IP
IP phone 1 IP phone 2
6
155789
PSTN
The initial OOD-R request can be authenticated and authorized using RFC 2617-based digest
authentication. To support authentication, Cisco Unified CME retrieves the credential information from
a text file stored in flash. This mechanism is used by Cisco Unified CME in addition to phone-based
credentials. The same credential file can be shared by other services that require request-based
authentication and authorization such as presence service. Up to five credential files can be configured
and loaded into the system. The contents of these five files are mutually exclusive, meaning the username
and password pairs must be unique across all the files. The username and password pairs must also be
different than those configured for SCCP or SIP phones in a Cisco Unified CME system.
For configuration information, see the “Enabling OOD-R” section on page 110.
Restrictions
Cisco Unified CME 3.4 and later versions support Media Flow-through mode only; enabling SIP-to-SIP
calls is required before you can successfully make SIP-to-SIP calls.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. allow-connections from-type to to-type
5. sip
6. registrar server [expires [max sec] [min sec]
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode and specifies Voice
over IP (VoIP) encapsulation.
Example:
Router(config)# voice service voip
Step 4 allow-connections from-type to to-type Enables calls between specific types of endpoints in a VoIP
network.
Example: • A separate allow-connections command is required for
Router(config-voi-srv)# allow-connections h323 each type of endpoint to be supported.
to h323
Router(config-voi-srv)# allow-connections h323
to SIP
Router(config-voi-srv)# allow-connections SIP
to SIP
Step 5 sip (Optional) Enters SIP configuration mode.
• Required if you are connecting IP phones running SIP
Example: directly in Cisco CME 3.4 and later.
Router(config-voi-srv)# sip
Example:
Router(config-voi-sip)# exit
Step 8 sip-ua Enters SIP user-agent configuration mode.
Example:
Router(config)# sip-ua
Step 9 notify telephone-event max-duration time Configures the maximum time interval allowed between
two consecutive NOTIFY messages for a single DTMF
event.
Example:
Router(config-sip-ua)# notify telephone-event • max-duration time—Range: 500 to 3000.
max-duration 2000 Default: 2000.
Step 10 registrar {dns:host-name | ipv4:ip-address} Registers E.164 numbers on behalf of analog telephone
expires seconds [tcp] [secondary] voice ports (FXS) and IP phone virtual voice ports (EFXS)
with an external SIP proxy or SIP registrar server.
Example:
Router(config-sip-ua)# registrar
ipv4:10.8.17.40 expires 3600 secondary
Step 11 retry register number Sets the total number of SIP Register messages that the
gateway should send.
Example: • number—Number of Register message retries.
Router(config-sip-ua)# retry register 10 Range: 1 to 10. Default: 10.
Defining DHCP
To set up DHCP service for your DHCP clients, perform only one of the following procedures:
• If your Cisco Unified CME router is the DHCP server and you can use a single shared address pool
for all your DHCP clients, see Defining a Single DHCP IP Address Pool, page 98.
• If your Cisco Unified CME router is the DHCP server and you need separate pools for each IP phone
and each non-IP-phone DHCP client, see Defining a Separate DHCP IP Address Pool for Each
DHCP Client, page 100.
• If the Cisco Unified CME router is not the DHCP server and you want to relay DHCP requests from
IP phones to a DHCP server on a different router, see Defining a DHCP Relay, page 102.
Note Do not perform this task if you already have a DHCP server on the LAN that can be used to provide
addresses to the Cisco Unified CME phones. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
Prerequisites
Restrictions
A single DHCP IP address pool cannot be used if non-IP-phone clients, such as PCs, must use a different
TFTP server address.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip dhcp pool pool-name
4. network ip-address [mask | /prefix-length]
5. option 150 ip ip-address
6. default-router ip-address
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ip dhcp pool pool-name Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.
Example:
Router(config)# ip dhcp pool mypool
Step 4 network ip-address [mask | /prefix-length] Specifies the IP address of the DHCP address pool
to be configured.
Example:
Router(config-dhcp)# network 10.0.0.0 255.255.0.0
Step 5 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco Unified IP phone downloads the image
configuration file.
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1 • This is your Cisco Unified CME router’s
address.
Step 6 default-router ip-address (Optional) Specifies the router that the IP phones
will use to send or receive IP traffic that is external
to their local subnet.
Example:
Router(config-dhcp)# default-router 10.0.0.1 • If the Cisco Unified CME router is the only
router on the network, this address should be the
Cisco Unified CME IP source address. This
command can be omitted if IP phones need to
send or receive IP traffic only to or from devices
on their local subnet.
• The IP address that you specify for default
router will be used by the IP phones for fallback
purposes. If the Cisco Unified CME IP source
address becomes unreachable, IP phones will
attempt to register to the address specified in
this command.
Step 7 end Returns to privileged EXEC mode.
Example:
Router(config-dhcp)# end
What to Do Next
• If you are configuring Cisco Unified CME for the first time on this router, you are ready to configure
NTP for the Cisco Unified CME router. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.
Note Do not perform this task if you already have a DHCP server on the LAN that can be used to provide
addresses to the Cisco Unified CME phones. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
Prerequisites
Restrictions
To use a separate DHCP IP address pool for each DHCP client, you must make an entry for every IP
phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip dhcp pool pool-name
4. host ip-address subnet-mask
5. client-identifier mac-address
6. option 150 ip ip-address
7. default-router ip-address
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ip dhcp pool pool-name Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.
Example:
Router(config)# ip dhcp pool pool2
Step 4 host ip-address subnet-mask Specifies the IP address that you want the phone to
get.
Example:
Router(config-dhcp)# host 10.0.0.0 255.255.0.0
Step 5 client-identifier mac-address Specifies the MAC address of the phone, which is
printed on a label on each Cisco Unified IP phone.
Example: • A separate client-identifier command is
Router(config-dhcp)# client-identifier 01238.380.3056 required for each DHCP client.
• Add “01” prefix number before the MAC
address.
Step 6 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco Unified IP phone downloads the image
configuration file.
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1 • This is your Cisco Unified CME router’s
address.
Step 7 default-router ip-address (Optional) Specifies the router that the IP phones
will use to send or receive IP traffic that is external
to their local subnet.
Example:
Router(config-dhcp)# default-router 10.0.0.1 • If the Cisco Unified CME router is the only
router on the network, this address should be
the Cisco Unified CME IP source address.
This command can be omitted if IP phones
need to send or receive IP traffic only to or
from devices on their local subnet.
• The IP address that you specify for default
router will be used by the IP phones for
fallback purposes. If the Cisco Unified CME
IP source address becomes unreachable, IP
phones will attempt to register to the address
specified in this command.
Example:
Router(config-dhcp)# end
What to Do Next
• If you are configuring Cisco Unified CME for the first time on this router, you are ready to configure
NTP for the Cisco Unified CME router. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.
Prerequisites
There is a DHCP server that is not on this Cisco Unified CME router on the LAN that can provide
addresses to the Cisco Unified CME phones.
Restrictions
SUMMARY STEPS
1. enable
2. configure terminal
3. service dhcp
4. interface type number
5. ip helper-address ip-address
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 service dhcp Enables the Cisco IOS DHCP server feature on the
router.
Example:
Router(config)# service dhcp
Step 4 interface type number Enters interface configuration mode for the
specified interface.
Example:
Router(config)# interface vlan 10
Step 5 ip helper-address ip-address Specifies the helper address for any unrecognized
broadcast for TFTP server and DNS server
requests.
Example:
Router(config-if)# ip helper-address 10.0.0.1 • A separate ip helper-address command is
required for each server if the servers are on
different hosts.
• You can also configure multiple TFTP server
targets by using the ip helper-address
commands for multiple servers.
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-if)# end
What to Do Next
• If you are configuring Cisco Unified CME for the first time on this router, you are ready to configure
NTP for the Cisco Unified CME router. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.
SUMMARY STEPS
1. enable
2. configure terminal
3. clock timezone zone hours-offset [minutes-offset]
4. clock summer-time zone recurring [week day month hh:mm week day month hh:mm [offset]]
5. ntp server ip-address
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 clock timezone zone hours-offset [minutes-offset] Sets the local time zone.
Example:
Router(config)# clock timezone pst -8
Step 4 clock summer-time zone recurring [week day month hh:mm (Optional) Specifies daylight savings time.
week day month hh:mm [offset]]
• Default: summer time is disabled. If the clock
summer-time zone recurring command is
Example: specified without parameters, the summer
Router(config)# clock summer-time pdt recurring time rules default to United States rules.
Default of the offset argument is 60.
Step 5 ntp server ip-address Synchronize software clock of router with the
specified NTP server.
Example:
Router(config)# ntp server 10.1.2.3
Step 6 exit Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
What to Do Next
• If you are configuring Cisco Unified CME for the first time on this router and if you have a multisite
installation, you are ready to configure a DTMF relay. See the “Configuring DTMF Relay for H.323
Networks in Multisite Installations” section on page 105.
• If Cisco Unified CME will interact with a SIP Gateway, you must set up support for the gateway.
See the Configuring SIP Trunk Support, page 106.
• If you are configuring Cisco Unified CME for the first time on this router and you are ready to
configure system parameters. See “Configuring System-Level Parameters” on page 119.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.
Note To configure DTMF relay on SIP networks, see the “Configuring SIP Trunk Support” on page 106.
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
4. dtmf-relay h245-alphanumeric
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 2 voip
Example:
Router(config-dial-peer)# end
What to Do Next
• To set up support for a SIP trunk, see the Configuring SIP Trunk Support, page 106.
• If you are configuring Cisco Unified CME for the first time on this router and you are ready to
configure system parameters. See “Configuring System-Level Parameters” on page 119.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
4. dtmf-relay rtp-nte
5. dtmf-relay sip-notify
6. exit
7. sip-ua
8. notify telephone-event max-duration msec
9. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]
10. retry register number
11. timers register msec
12. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 2 voip
Step 4 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event (NTE)
payload type and enables DTMF relay using the RFC 2833
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte
standard method.
Step 5 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages.
Example:
Router(config-dial-peer)# dtmf-relay sip-notify
Step 6 exit Exits dial-peer configuration mode.
Example:
Router(config-dial-peer)# exit
Step 7 sip-ua Enters SIP user-agent configuration mode.
Example:
Router(config)# sip-ua
Step 8 notify telephone-event max-duration msec Sets the maximum milliseconds allowed between two
consecutive NOTIFY messages for a single DTMF event.
Example: • max-duration time—Range: 500 to 3000.
Router(config-sip-ua)# notify telephone-event Default: 2000.
max-duration 2000
Step 9 registrar {dns:host-name | ipv4:ip-address} Registers E.164 numbers on behalf of analog telephone
expires seconds [tcp] [secondary] voice ports (FXS) and IP phone virtual voice ports (EFXS)
with an external SIP proxy or SIP registrar server.
Example:
Router(config-sip-ua)# registrar
ipv4:10.8.17.40 expires 3600 secondary
Step 10 retry register number Sets the total number of SIP Register messages that the
gateway should send.
Example: • number—Number of Register message retries.
Router(config-sip-ua)# retry register 10 Range: 1 to 10. Default: 10.
Example:
Router(config-sip-ua)# end
SUMMARY STEPS
DETAILED STEPS
Prerequisites
Your Cisco Unified CME router is a DHCP server.
Restrictions
If the DHCP server is on a different router than Cisco Unified CME, reconfigure the external DHCP
server with the new IP address of the TFTP server.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip dhcp pool pool-name
4. option 150 ip ip-address
5. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-dhcp)# end
Enabling OOD-R
To enable OOD-R support on the Cisco Unified CME router, perform the following steps.
Prerequisites
• Cisco Unified CME 4.1 or a later version.
• Cisco IOS Release 12.4(11)XJ or a later release.
• The application that initiates OOD-R, such as a click-to-dial application, and its directory server
must be installed and configured.
– For information on the SIP REFER and NOTIFY methods used between the directory server
and Cisco Unified CME, see RFC 3515, The Session Initiation Protocol (SIP) Refer Method.
– For information on the message flow Cisco Unified CME uses when initiating a session
between the Referee and Refer-Target, see RFC 3725, Best Current Practices for Third Party
Call Control (3pcc).
Restrictions
• The call waiting, conferencing, hold, and transfer call features are not supported while the
Refer-Target is ringing.
• In a SIP to SIP scenario, no ringback is heard by the Referee when Refer-Target is ringing.
SUMMARY STEPS
1. enable
2. configure terminal
3. sip-ua
4. refer-ood enable [request-limit]
5. exit
6. voice register global
7. authenticate ood-refer
8. authenticate credential tag location
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 sip-ua Enters SIP user-agent configuration mode to configure the
user agent.
Example:
Router(config)# sip-ua
Step 4 refer-ood enable [request-limit] Enables OOD-R processing.
• request-limit—Maximum number of concurrent
Example: incoming OOD-R requests that the router can process.
Router(config-sip-ua)# refer-ood enable 300 Range: 1 to 500. Default: 500.
Step 5 exit Exits SIP user-agent configuration mode.
Example:
Router(config-sip-ua)# exit
Step 6 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME or Cisco Unified SRST environment.
Example:
Router(config)# voice register global
Step 7 authenticate ood-refer (Optional) Enables authentication of incoming OOD-R
requests using RFC 2617-based digest authentication.
Example:
Router(config-register-global)# authenticate
ood-refer
Example:
Router(config-register-global)# end
Troubleshooting OOD-R
Step 1 debug ccsip messages
This command displays the SIP messages exchanged between the SIP UA client and the router.
Router# debug ccsip messages
OOD-R: Example
!
voice register global
mode cme
source-address 11.1.1.2 port 5060
load 7971 SIP70.8-0-1-11S
load 7970 SIP70.8-0-1-11S
load 7961GE SIP41.8-0-1-0DEV
load 7961 SIP41.8-0-1-0DEV
authenticate ood-refer
authenticate credential 1 tftp://172.18.207.15/labtest/cred1.csv
create profile sync 0004550081249644
.
.
.
sip-ua
authentication username jack password 021201481F
refer-ood enable
!
Where to Go Next
• If you are configuring Cisco Unified CME for the first time on this router, you are ready to
configure system-level parameters. See “Configuring System-Level Parameters” on page 119.
• If you modified network parameters for an already configured Cisco Unified CME router, you are
ready to generate the configuration file to save the modifications. See “Generating Configuration
Files for Phones” on page 245
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 6 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the system-level settings to configure before you add devices and configure
Cisco Unified Communications Manager Express (Cisco Unified CME) features.
Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip this module.
Contents
• Prerequisites for System-Level Parameters, page 119
• Information About Configuring System-Level Parameters, page 120
• How to Configure System-Level Parameters, page 122
• Configuration Examples for System-Level Parameters, page 141
• Where to Go Next, page 143
• Additional References, page 143
• Feature Information for System-Level Parameters, page 145
Note When the storage location chosen is flash and the file system type on this device is Class B
(LEFS), check free space on the device periodically and use the squeeze command to free the
space used up by deleted files. Unless you use the squeeze command, the space used by the
moved or deleted configuration files cannot be used by other files. Rewriting flash memory
space during the squeeze operation may take several minutes. We recommend using this
command during scheduled maintenance periods or off-peak hours.
• TFTP—When an external TFTP server is the storage location, you can create additional
configuration files that can be applied per phone type or per individual phone. Up to five user and
network locales can be used in these configuration files. To store configuration files on an external
TFTP server, use the cnf-file location tftp url command.
You can then specify one of the following ways to create configuration files:
• Per system—This is the default. All phones use a single configuration file. The default user and
network locale in a single configuration file are applied to all phones in the Cisco Unified CME
system. Multiple locales and user-defined locales are not supported. To use the per-system option,
either do not use the cnf-file command or use the no cnf-file command to reset the option from a
different configuration.
• Per phone type—This setting creates separate configuration files for each phone type. For example,
all Cisco Unified IP Phone 7960s use XMLDefault7960.cnf.xml, and all Cisco Unified IP
Phone 7905s use XMLDefault7905.cnf.xml. All phones of the same type use the same configuration
file, which is generated using the default user and network locale. To create configuration files per
phone type, use the cnf-file perphonetype command. This option is not supported if you store the
configuration files in the system location.
• Per phone—This setting creates a separate configuration file for each phone, by MAC address. For
example, a Cisco Unified IP Phone 7960 with the MAC address 123.456.789 creates the per-phone
configuration file SEP123456789.cnf.xml. The configuration file for a phone generates with the
default user and network locale unless a different user and network locale is applied to the phone
using an ephone template. To create configuration files per phone type, use the cnf-file perphone
command. This option is not supported if you store the configuration files in the system location.
For configuration information, see the “SCCP: Defining Per-Phone Configuration Files and Alternate
Location” section on page 129.
presented to both the primary and secondary Cisco Unified CME routers. The primary router is
configured by default to answer the call immediately. The secondary Cisco Unified CME router is
configured to answer the call after three rings using the voice-port ring number 3 command. If the
primary router is operational, it answers the call immediately and changes the call state so that the
secondary router does not try to answer it. If the primary router is unavailable and does not answer the
call, the secondary router sees the new call coming in and answers after three rings.
The secondary Cisco Unified CME router should be connected in some way on the LAN, either through
the same switch or through another switch that may or may not be connected to the primary
Cisco Unified CME router directly. As long as both routers and the phones are connected on the LAN
with the appropriate configurations in place, the phones can register to whichever router is active.
Configure primary and secondary Cisco Unified CME routers identically, with the exception that the
FXO voice port from the PSTN on the secondary router should be configured to answer after more rings
than the primary router, as previously explained. The ip source-address command is used on both
routers to specify the IP addresses of the primary and secondary routers.
Timeouts
The following system-level timeout parameters have default values that are generally adequate:
• Busy Timeout—Amount of time that can elapse after a transferred call reaches a busy signal before
the call is disconnected.
• Interdigit Timeout—Amount of time that can elapse between the receipt of individual dialed digits
before the dialing process times out and is terminated. If the timeout ends before the destination is
identified, a tone sounds and the call ends. This value is important when using variable-length
dial-peer destination patterns (dial plans). For more information, see Dial Peer Configuration on
Voice Gateway Routers.
• Ringing Timeout—Amount of time a phone can ring with no answer before returning a disconnect
code to the caller. This timeout is used only for extensions that do not have no-answer call
forwarding enabled. The ringing timeout prevents hung calls received over interfaces such as FXO
that do not have forward-disconnect supervision.
• Keepalive—Interval determines how often a message is sent between the router and
Cisco Unified IP phones, over the session to ensure that the keepalive timeout is not exceeded. If no
other traffic is sent over the session during the interval, a keepalive message is sent.
SCCP
• SCCP: Setting Up Cisco Unified CME, page 125 (required)
• SCCP: Setting Date and Time Parameters, page 127 (required)
• SCCP: Blocking Automatic Registration, page 128 (optional)
• SCCP: Defining Per-Phone Configuration Files and Alternate Location, page 129 (optional)
• SCCP: Changing Defaults for Time Outs, page 130 (optional)
• SCCP: Configuring a Redundant Router, page 132 (optional)
SIP
• SIP: Setting Up Cisco Unified CME, page 135 (required)
• SIP: Setting Date and Time Parameters, page 137 (required)
• SIP: Setting Network Time Protocol, page 139 (required)
• SIP: Changing Session-Level Application for SIP Phones, page 140 (optional)
Note To specify that an individual directory number not register with the external registrar by using this
command. For configuration information, see the “SIP: Disabling SIP Proxy Registration for a Directory
Number” section on page 172.
Prerequisites
Cisco Unified CME 3.4 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. mode cme
5. bulk number
6. exit
7. sip-ua
8. registrar {dns:address | ipv4:destination-address} expires seconds [tcp] [secondary ] no
registrar [secondary ]
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 5 bulk number Sets bulk registration for E.164 numbers that will register
with SIP proxy server.
Example: • number—Unique sequence of up to 32 characters
Router(config-register-global)# bulk 408526.... including wild cards and patterns that represents E.164
n umbers that will register with Sip proxy server.
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-pool)# exit
Step 7 sip-ua Enters Session Initiation Protocol (SIP) user agent (ua)
configuration mode for configuring the user agent.
Example:
Router(config)# sip-ua
Step 8 registrar {dns:address | Enables SIP gateways to register E.164 numbers with SIP
ipv4:destination-address} expires seconds [tcp] proxy server.
[secondary] no registrar [secondary]
Example:
Router(config-sip-ua)# registrar server
ipv4:1.5.49.240
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-sip-ua)# end
Examples
The following example shows that all phone numbers that match the pattern “408555..” can register with
a SIP proxy server (IP address 1.5.49.240):
voice register global
mode cme
bulk 408555….
sip-ua
registrar ipv4:1.5.49.240
SUMMARY STEPS
1. enable
2. configure terminal
3. tftp-server flash:filename
4. telephony-service
5. load phone-type firmware-file
6. max-ephones max-phones
7. max-dn max-directory-numbers [preference preference-order] [no-reg primary | both]
8. ip source-address ip-address port port [any-match | strict-match]
9. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# telephony-service
Step 5 load phone-type firmware-file Identifies a Cisco Unified IP phone firmware file to be
used by phones of the specified type when they register.
Example: • A separate load command is required for each phone
Router(config-telephony)# load 7960-7940 type.
P00307020300
Note If you are loading a firmware file larger than
384 KB, you must first load a file for that phone
type that is smaller than 384 KB, then load the
larger file.
Step 6 max-ephones max-phones Limits number of phones to be supported by this router.
• Maximum number is platform and version-specific.
Example: Type? for value.
Router(config-telephony)# max-ephones 24
Step 7 max-dn max- directory- numbers [preference Limits number of directory numbers to be supported by
preference-order] [no-reg primary | both] this router.
• Maximum number is platform and version-specific.
Example: Type? for value.
Router(config-telephony)# max-dn 200 no-reg
primary
Step 8 ip source-address ip-address [port port] [any-match Identifies the IP address and port number that the
| strict-match] Cisco Unified CME router uses for IP phone registration.
• port port—(Optional) TCP/IP port number to use
Example: for SCCP. Range is 2000 to 9999. Default is 2000.
Router(config-telephony)# ip source-address
10.16.32.144 • any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• strict-match—(Optional) Instructs the router to
reject IP phone registration attempts if the IP server
address used by the phone does not exactly match
the source address.
Step 9 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd}
5. time-format {12 | 24}
6. time-zone number
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | (Optional) Sets the date format for phone display.
yy-mm-dd}
• Default: mm-dd-yy.
Example:
Router(config-telephony)# date-format yy-dd-mm
Step 5 time-format {12 | 24} (Optional) Selects a 12 or 24-hour clock for the time
display format on phone display.
Example: • Default: 12.
Router(config-telephony)# time-format 24
Example:
Router(config-telephony)# end
Prerequisite
Cisco Unified CME 4.0 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. auto-reg-ephone
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
Prerequisites
Cisco Unified CME 4.0 or a later version.
Restrictions
• Externally stored and per-phone configuration files are not supported on the Cisco Unified IP Phone
7902G, 7910, 7910G, or 7920, or the Cisco Unified IP Conference Station 7935 and 7936.
• TFTP does not support file deletion. When configuration files are updated, they overwrite any
existing configuration files with the same name. If you change the configuration file location, files
are not deleted from the TFTP server.
• Generating configuration files on flash or slot 0 can take up to a minute, depending on the number
of files being generated.
• For smaller routers such as Cisco 2600 series routers, you must manually enter the squeeze
command to erase files after changing the configuration file location or entering any commands that
trigger the deletion of configuration files. Unless you use the squeeze command, the space used by
the moved or deleted configuration files is not usable by other files.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. cnf-file location {flash: | slot0: | tftp tftp-url}
5. cnf-file {perphonetype | perphone}
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 cnf-file location {flash: | slot0: | tftp Specifies a location other than system:/its for storing
tftp-url} phone configuration files.
• Required for per-phone or per-phone type
Example: configuration files.
Router(config-telephony)# cnf-file location
flash:
Step 5 cnf-file {perphonetype | perphone} Specifies whether to use a separate file for each type of
phone or for each individual phone.
Example:
Router(config-telephony)# cnf-file perphone
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
What to Do Next
If you changed the configuration file storage location, use the option 150 ip command to update the
address. See “Changing the TFTP Address on a DHCP Server” on page 109.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. timeouts busy seconds
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 timeouts busy seconds (Optional) Sets the amount of time after which calls that are
transferred to busy destinations are disconnected.
Example: • seconds—Number of seconds. Range is 0 to 30. Default
Router(config-telephony)# timeouts busy 20 is 10.
Step 5 timeouts interdigit seconds (Optional) Configures the interdigit timeout value for all
Cisco Unified IP phones attached to the router.
Example: • seconds—Number of seconds before the interdigit
Router(config-telephony)# timeouts interdigit timer expires. Range is 2 to 120. Default is 10.
30
Step 6 timeouts ringing seconds (Optional) Sets the duration, in seconds, for which the
Cisco Unified CME system allows ringing to continue if a
call is not answered. Range is 5 to 60000. Default is 180.
Example:
Router(config-telephony)# timeouts ringing 30
Step 7 keepalive seconds (Optional) Sets the time interval, in seconds, between
keepalive messages that are sent to the router by
Cisco Unified IP phones.
Example:
Router(config-telephony)# keepalive 45 • The default is usually adequate. If the interval is set too
large, it is possible that notification will be delayed
when a system goes down.
• Range: 10 to 65535. Default: 0.
Step 8 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Prerequisites
• Cisco Unified CME 4.0 or a later version.
• The secondary router must have a running configuration identical to that of the primary router.
• The physical configuration of the secondary router must be as described in the “Redundant Cisco
Unified CME Router” section on page 121.
• Phones that use this feature must be configured with the type command, which guarantees that the
appropriate phone configuration file will be present.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. ip source-address ip-address port port [secondary ip-address [rehome seconds]] [any-match |
strict-match]
5. exit
6. voice-port slot-number/port
7. signal ground-start
8. incoming alerting ring-only
9. ring number number
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# exit
Step 6 voice-port slot-number/port Enters voice-port configuration mode for the FXO voice
port for DID calls from the PSTN.
Example:
Router(config)# voice-port 2/0
Step 7 signal ground-start Specifies ground-start signaling for a voice port.
Example:
Router(config-voiceport)# signal ground-start
Step 8 incoming alerting ring-only Instructs the FXO ground-start voice port to detect
incoming calls by detecting incoming ring signals.
Example:
Router(config-voiceport)# incoming alerting
ring-only
Example:
Router(config-telephony)# exit
Step 6 voice-port slot-number/port Enters voice-port configuration mode for the FXO voice
port for DID calls from the PSTN.
Example:
Router(config)# voice-port 2/0
Step 7 signal ground-start Specifies ground-start signaling for a voice port.
Example:
Router(config-voiceport)# signal ground-start
Step 8 incoming alerting ring-only Instructs the FXO ground-start voice port to detect
incoming calls by detecting incoming ring signals.
Example:
Router(config-voiceport)# incoming alerting
ring-only
Example:
Router(config-voiceport)# end
Note If your Cisco Unified CME system supports SCCP and SIP phones, do not connect your SIP phones to
your network until after you have verified the configuration profile for the SIP phone.
Prerequisites
• Cisco CME 3.4 or a later version.
Restrictions
Java-based phones, such as the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE, are supported only in Cisco Unified CME 4.1 and later.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. mode cme
5. source-address ip-address
6. load phone-type firmware-file
7. tftp path {system: | flash: | slot0: | tftp tftp-url}
8. max-pool max-phones
9. max-dn max-directory-numbers
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 5 source-address ip-address [port port] Enables the Cisco Unified CME router to receive messages
from SIP phones through the specified IP address and port.
Example: • port—(Optional) TCP/IP port number.
Router(config-register-global)# source-address Range: 2000 to 9999. Default: 2000.
10.6.21.4
Step 6 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each phone
Example: type.
Router(config-register-global)# load 7960-7940
P0S3-07-3-00
Step 7 tftp-path {system: | flash: | slot0: | tftp: Defines the location from which the SIP phones will
tftp-url} download configuration profile files.
• Default: system.
Example:
Router(config-register-global)# tftp-path
http://mycompany.com/files
Step 8 max-pool max-phones Limits number of SIP phones to be supported by the
Cisco Unified CME router.
Example: • Default: 0.
Router(config-register-global)# max-pool 10
Step 9 max-dn max-directory-numbers (Optional) Limits number of directory numbers for SIP
phones to be supported by the Cisco Unified CME router.
Example: • Default: 150 or maximum allowed on platform, by
Router(config-register-global)# max-dn 20 version. Type ? for value.
Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command is enabled.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. timezone number
5. date-format [d/m/d | m/d/y | y-d-m | y/d/m | y/m/d | yy-m-d]
DETAILED STEPS
Example:
Router# configure terminal
Prerequisites
• Cisco Unified CME 4.1 or a later version.
• The firmware load 8.2(1) or a later version is installed for SIP phones to download. For upgrade
information, see the “SIP: Upgrading or Downgrading Phone Firmware Between Versions” section
on page 77.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. ntp-server ip-address [mode {anycast | directedbroadcast | multicast | unicast}]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to
set global parameters for all supported SIP phones
in a Cisco Unified CME environment.
Example:
Router(config)# voice register global
Step 4 ntp-server ip-address [mode {anycast | Clock on this router is synchronized with the
directedbroadcast | multicast | unicast}] specified NTP server.
Example:
Router(config-register-global)# ntp-server 10.1.2.3
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-register-global)# end
Prerequisites
• Cisco CME 3.4 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. application application-name
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 application application-name (Optional) Changes the default application for all dial peers
associated with the SIP phones in Cisco Unified CME to the
specified application.
Example:
Router(config-register-global)# application Note The application command in voice register pool
sipapp2 configuration mode takes precedence over this
command in voice register global configuration
mode.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
Cisco Unified IP Phone 7911, 7941, 7941-GE, 7961, 7961-GE, 7970, and 7971 require multiple files to
be shared using TFTP. The following configuration example adds support for these phones.
tftp-server flash:SCCP11.7-2-1-0S.loads
tftp-server flash:term11.default.loads
tftp-server flash:apps11.1-0-0-72.sbn
tftp-server flash:cnu11.3-0-0-81.sbn
tftp-server flash:cvm11.7-2-0-66.sbn
tftp-server flash:dsp11.1-0-0-73.sbn
tftp-server flash:jar11.7-2-0-66.sbn
! 7911 firmware
!
tftp-server flash:TERM41.7-0-3-0S.loads
tftp-server flash:TERM41.DEFAULT.loads
tftp-server flash:TERM61.DEFAULT.loads
tftp-server flash:CVM41.2-0-2-26.sbn
tftp-server flash:cnu41.2-7-6-26.sbn
tftp-server flash:Jar41.2-9-2-26.sbn
! 7941/41-GE, 7961/61-GE firmware
!
tftp-server flash:TERM70.7-0-1-0s.LOADS
tftp-server flash:TERM70.DEFAULT.loads
tftp-server flash:TERM71.DEFAULT.loads
tftp-server flash:CVM70.2-0-2-26.sbn
tftp-server flash:cnu70.2-7-6-26.sbn
tftp-server flash:Jar70.2-9-2-26.sbn
! 7970/71 firmware
!
telephony-service
load 7911 SCCP11.7-2-1-0S
load 7941 TERM41.7-0-3-0S
load 7961 TERM41.7-0-3-0S
load 7941GE TERM41.7-0-3-0S
load 7961GE TERM41.7-0-3-0S
load 7970 TERM70.7-0-1-0s
load 7971 TERM70.7-0-1-0s
create cnf-files version-stamp Jan 01 2002 00:00:00
.
.
.
voice-port 3/0/0
signal ground-start
incoming alerting ring-only
The secondary Cisco Unified CME router is configured with the same commands, except that the ring
number command is set to 3 instead of using the default of 1.
telephony-service
ip source-address 10.0.0.1 port 2000 secondary 10.5.2.78
voice-port 3/0/0
signal ground-start
incoming alerting ring-only
ring number 3
Where to Go Next
After configuring system-level parameters, you are ready to configure phones in Cisco Unified CME for
making basic calls. See “Configuring Phones to Make Basic Calls” on page 147.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 7 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This module describes how to configure Cisco Unified IP phones in a Cisco Unified Communications
Manager Express (Cisco Unified CME) system so that you can make and receive basic calls.
Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip this module unless you want to modify the configuration to add phones.
Contents
• Prerequisites for Configuring Phones to Make Basic Calls, page 148
• Restrictions for Configuring Phones to Make Basic Calls, page 148
• Information About Configuring Phones to Make Basic Calls, page 148
• How to Configure Phones for a PBX System, page 158
• How to Configure Phones for a Key System, page 176
• How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator,
page 188
• Configuration Examples for Making Basic Calls, page 199
• Where to Go Next, page 208
• Additional References, page 209
• Feature Information for Configuring Phones to Make Basic Calls, page 210
Directory Numbers
A directory number, also known as an ephone-dn for SCCP or a voice-register dn for SIP, is the software
configuration in Cisco Unified CME that represents the line connecting a voice channel to a phone. A
directory number has one or more extension or telephone numbers associated with it to allow call
connections to be made. Generally, a directory number is equivalent to a phone line, but not always.
There are several types of directory numbers, which have different characteristics.
Each directory number has a unique dn-tag, or sequence number, to identify it during configuration.
Directory numbers are assigned to line buttons on phones during configuration.
One virtual voice port and one or more dial peers are automatically created for each directory number,
depending on the configuration for SCCP phones, or for SIP phones, when the phone registers in
Cisco Unified CME.
The number of directory numbers that you create corresponds to the number of simultaneous calls that
you can have, because each directory number represents a virtual voice port in the router. This means
that if you want more than one call to the same number to be answered simultaneously, you need multiple
directory numbers with the same destination number pattern.
The directory number is the basic building block of a Cisco Unified CME system. Six different types of
directory number can be combined in different ways for different call coverage situations. Each type will
help with a particular type of limitation or call-coverage need. For example, if you want to keep the
number of directory numbers low and provide service to a large number of people, you might use shared
directory numbers. Or if you have a limited quantity of extension numbers that you can use and you need
to have a large quantity of simultaneous calls, you might create two or more directory numbers with the
same number. The key is knowing how each type of directory number works and its advantages.
Not all types of directory numbers can be configured for all phones or for all protocols. In the remaining
information about directory numbers, we have used SCCP in the examples presented but that does not
imply exclusivity. The following sections describe the types of directory number in a
Cisco Unified CME system:
• Single-Line, page 149
• Dual-Line, page 150
• Two Directory Numbers with One Telephone or Extension Number, page 151
• Dual-Number, page 152
• Shared, page 152
• Monitor Mode for Shared Lines, page 153
• Overlaid, page 153
Single-Line
A single-line directory number has the following characteristics:
• Makes one call connection at a time using one phone line button. A single-line directory number has
one telephone number associated with it.
• Should be used when phone buttons have a one-to-one correspondence to the PSTN lines that come
into a Cisco Unified CME system.
• Should be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI),
loopback, and music-on-hold (MOH) feed sources.
• When used with multiple-line features like call waiting, call transfer, and conferencing, there must
be more than one single-line directory number on a phone.
• Can be combined with dual-line directory numbers on the same phone.
Note that you must make the choice to configure each directory number in your system as either dual-line
or single-line when you initially create configuration entries. If you need to change from single-line to
dual-line later, you must delete the configuration for the directory number, then recreate it.
Figure 7 shows a single-line directory number for an SCCP phone in Cisco Unified CME.
ephone-dn 11
number 1001
IP V ephone 1
88888
Phone 1 button 1:11
Button 1 is extension 1001
Dual-Line
A dual-line directory number has the following characteristics:
• One voice port with two channels.
• Supported on IP phones that are running SCCP; not supported on IP phones that are running SIP.
• Can make two call connections at the same time using one phone line button. A dual-line directory
number has two channels for separate call connections.
• Can have one number or two numbers (primary and secondary) associated with it.
• Should be used for a directory number that needs to use one line button for features like call waiting,
call transfer, or conferencing.
• Cannot be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI),
loopback, and music-on-hold (MOH) feed sources.
• Can be combined with single-line directory numbers on the same phone.
Note that you must make the choice to configure each directory number in your system as either dual-line
or single-line when you initially create configuration entries. If you need to change from single-line to
dual-line later, you must delete the configuration for the directory number, then recreate it.
Figure 8 shows a dual-line directory number for an SCCP phone in Cisco Unified CME.
ephone-dn 12 dual-line
number 1002
IP V ephone 2
88889
ephone-dn 13
number 1003
no huntstop
ephone-dn 14
IP V number 1003
Phone 3 preference 1
Button 1 is extension 1003
88891
Phone 4 ephone-dn 13
Button 1 is extension 1003 number 1003
no huntstop
IP
ephone-dn 14
number 1003
IP V preference 1
Phone 5 ephone 4
Button 1 is extension 1003 button 1:13
88892
ephone 5
button 1:14
Dual-Number
A dual-number directory number has the following characteristics:
• Has two telephone numbers, a primary number and a secondary number.
• Can make one call connection if it is a single-line directory number.
• Can make two call connections at a time if it is a dual-line directory number (SCCP only).
• Should be used when you want to have two different numbers for the same button without using
more than one directory number.
Figure 11 shows a directory number that has two numbers, extension 1006 and extension 1007.
ephone-dn 15
number 1006 secondary 1007
IP V
ephone 6
88890
Phone 6 button 1:15
Button 1 is extension 1006
Button 1 is also extension 1007
Shared
A shared directory number has the following characteristics:
• Line appears on two different phones but uses the same directory number, and extension or phone
number.
• Can make one call at a time and that call appears on both phones.
• Should be used when you want the capability to answer or pick up a call at more than one phone.
Because these phones share the same directory number, if the directory number is connected to a call on
one phone, that directory number is unavailable for other calls on the second phone. If a call is placed
on hold on one phone, it can be retrieved on the second phone. This is like having a single-line phone in
your house with multiple extensions. You can answer the call from any phone on which the number
appears, and you can pick it up from hold on any phone on which the number appears.
Figure 12 shows a shared directory number on phones that are running SCCP. Extension 1008 appears
on both phone 7 and phone 8.
Phone 7
Button 1 is extension 1008
ephone-dn 16
number 1008
IP
ephone 7
button 1:16
IP V
Phone 8 ephone 8
88893
Button 1 is extension 1008 button 1:16
Overlaid
An overlaid directory number has the following characteristics:
• Is a member of an overlay set, which includes all the directory numbers that have been assigned
together to a particular phone button.
• Can have the same telephone or extension number as other members of the overlay set or different
numbers.
• Can be single-line or dual-line, but cannot be mixed single-line and dual-line in the same overlay set.
• Can be shared on more than one phone.
Overlaid directory numbers provide call coverage similar to shared directory numbers because the same
number can appear on more than one phone. The advantage of using two directory numbers in an overlay
arrangement rather than as a simple shared line is that a call to the number on one phone does not block
the use of the same number on the other phone, as would happen if it were a shared directory number.
For information about configuring call coverage using overlaid ephone-dns, see “Configuring
Call-Coverage Features” on page 563.
You can overlay up to 25 lines on a single button. A typical use of overlaid directory numbers would be
to create a “10x10” shared line, with ten lines in an overlay set shared by ten phones, resulting in the
possibility of ten simultaneous calls to the same number. For configuration information, see the “SCCP:
Creating Directory Numbers for a Simple Key System” section on page 176
Analog Phones
Cisco Unified CME supports analog phones using Cisco Analog Telephone Adaptors (ATAs) or FXS
ports in SCCP mode or H.323 mode, and supports fax machines on Cisco ATA or FXS ports in H.323
mode. The FXS ports used for analog phones or fax can be on the Cisco Unified CME router or on a
Cisco VG 224 voice gateway or Integrated Services Router (ISR). This section provides information on
the following topics:
• Cisco ATAs in SCCP Mode, page 155
• FXS Ports in SCCP Mode, page 155
• FXS Ports in H.323 Mode, page 155
– Configure an IPsec VPN tunnel between the remote site router (or example, a Cisco 831) and
the Cisco Unified CME router. This solution requires an Advanced IP Services or higher image
on the Cisco Unified CME router if this router is used to terminate the VPN tunnel. Voice will
be encrypted across the WAN. This method will also work with the Cisco VPN client on a PC
to support Cisco IP Communicator.
IP WAN PSTN
146625
Teleworker Cisco 831
remote phone NAT firewall Cisco Unified CME
router (VPN)
router
• Cisco Unified IP Phone 7905 and 7912—The dial plan is a field in their configuration files.
• Cisco Unified IP Phone 7911G, 7940, 7941G, 7941GE, 7960, 7961G, 7961GE, 7970G, and
7971GE—The dial plan is a separate XML file that is pointed to from the normal configuration file.
For configuration information for Cisco Unified CME, see the “SIP: Configuring Dial Plans” section on
page 165.
Note To create and assign directory numbers to be included in an overlay set, see “SCCP: Configuring
Overlaid Ephone-dns” on page 615.
Prerequisites
• The maximum number of directory numbers must be configured for other than the default, by using
the max-dn command.
Restrictions
• The Cisco Unified IP Phone 7931G is a SCCP keyset phone and when configured for a key system,
does not support the dual-line option for a directory number. To configure a Cisco Unified IP Phone
7931G, see the “How to Configure Phones for a Key System” section on page 176.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. name name
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • Configuring a dual-line supports features such as call
Router(config)# ephone-dn 55 dual-line waiting, call transfer, and conferencing with a single
ephone-dn.
• To change an ephone-dn from dual-line to single-line
mode or the reverse, you must first delete the
ephone-dn and then recreate it.
Step 4 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn
| primary]] instance.
• Configuring a secondary number supports features
Example: such as call waiting, call transfer, and conferencing
Router(config-ephone-dn)# number 2345 with a single ephone-dn.
Example:
Router(config-telephony)# end
What to Do Next
After creating directory numbers, you can assign one or more directory number to a Cisco Unified IP
phone. See “SCCP: Assigning Directory Numbers to Phones” section on page 160.
Note To create and assign directory numbers to be included in an overlay set, see “SCCP: Configuring
Overlaid Ephone-dns” on page 615.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address [mac-address]
5. type phone-type [addon 1 module-type [2 module-type]]
6. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]
7. keypad-normalize
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks. The maximum
Router(config)# ephone 6 number of ephones is version and platform-specific.
Type ? to display range.
Step 4 mac-address [mac-address] Specifies the MAC address of the IP phone that is being
configured.
Example: • mac-address—(Optional) For Cisco Unified CME 3.0
Router(config-ephone)# mac-address 2946.3f2.311 and later, not required to register phones before
configuring the phone because Cisco Unified CME
can detect MAC addresses and automatically populate
phone configurations with the MAC addresses and
phone types for individual phones. Not supported for
voice-mail ports.
Step 5 type phone-type [addon 1 module-type Specifies the type of phone.
[2 module-type]]
• Cisco Unified CME 4.0 and later versions—The only
types to which you can apply an add-on module are
Example: 7960, 7961, 7961GE, and 7970.
Router(config-ephone)# type 7960 addon 1 7914
• Cisco CME 3.4 and earlier versions—The only type to
which you can apply an add-on module is 7960.
Step 6 button button-number{separator}dn-tag Associates a button number and line characteristics with an
[,dn-tag...] extension (ephone-dn). Maximum number of buttons is
[button-number{x}overlay-button-number]
[button-number...]
determined by phone type.
Note The Cisco Unified IP Phone 7910 has only one line
button, but can be given two ephone-dn tags.
Example:
Router(config-ephone)# button 1:10 2:11 3b12
4o13,14,15
Step 7 keypad-normalize (Optional) Imposes a 200-millisecond delay before each
keypad message from an IP phone.
Example:
Router(config-ephone)# keypad-normalize
Step 8 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.
Examples
The following example assigns extension 2225 in the Accounting Department to button 1 on ephone 2.
ephone-dn 25
number 2225
name Accounting
ephone 2
mac-address 00E1.CB13.0395
type 7960
button 1:25
Prerequisites
• Cisco CME 3.4 or a later version.
• The maximum number of directory numbers supported by a router is version and platform
dependent. To configure more directory numbers than the default, use the max-dn (voice register
global) command before performing this procedure. For configuration information, see “SIP:
Setting Up Cisco Unified CME” on page 135.
Restrictions
• Call forward all, presence, and message-waiting indication (MWI) features in Cisco Unified CME
4.1 and later versions require that SIP phones are configured with a directory number (using dn
keyword in number command); direct line numbers are not supported.
• Shared lines are not supported by SIP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config)# voice register dn 17
Step 4 number number Defines a valid number for a directory number.
Example:
Router(config-register-dn)# number 7001
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end
Note If your Cisco Unified CME system supports SCCP and also SIP phones, do not connect your SIP phones
to your network until after you have verified the configuration profile for the SIP phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. id mac address
5. type phone-type
6. number tag dn dn-tag
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 id {network address mask mask | ip address mask Explicitly identifies a locally available individual SIP phone to
mask | mac address} support a degree of authentication.
Example:
Router(config-register-pool)# id mac
0009.A3D4.1234
Step 5 type phone-type Defines a phone type for the SIP phone being configured.
Example:
Router(config-register-pool)# type 7960-7940
Step 6 number tag dn dn-tag Associates a directory number with the SIP phone being
configured.
Example: • dn dn-tag—Identifies the directory number for this SIP
Router(config-register-pool)# number 1 dn 17 phone as defined by the voice register dn command.
Step 7 username username password string (Optional) Required only if authentication is enabled with
the authenticate command. Creates an authentication
credential.
Example:
Router(config-register-pool)# username smith Note This command is not for SIP proxy registration. The
password 123zyx password will not be encrypted. All lines in a phone
will share the same credential.
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• If you want to select the session-transport protocol for a SIP phone, see the “SIP: Selecting
Session-Transport Protocol for a Phone” section on page 171.
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones”
on page 250.
Prerequisites
• Cisco Unified CME 4.1 or a later version.
• mode cme command must be enabled in Cisco Unified CME.
Restrictions
• If you create a dial plan by downloading a custom XML dial pattern file to flash and using the
filename command, and the XML file contains an error, the dial plan might not work properly on a
phone. We recommend creating a dial pattern file using the pattern command.
• To remove a dial plan that was created using a custom XML file with the filename command, you
must remove the dial plan from the phone, create a new configuration profile, and then use the reset
command to reboot the phone. You can use the restart command after removing a dial plan from a
phone only if the dial plan was created using the pattern command.
• To use KPML if a matching dial plan is not found, when both a dial plan and KPML are enabled on
a phone, you must configure a dial pattern with a single wildcard character (.) as the last pattern in
the dial plan. For example:
voice register dialplan 10
type 7940-7960-others
pattern 1 66...
pattern 2 91.......
pattern 3 .
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dialplan dialplan-tag
4. type phone-type
5. pattern tag string [button button-number] [timeout seconds] [user {ip | phone}]
or
filename filename
6. exit
7. voice register pool pool-tag
8. dialplan dialplan-tag
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dialplan dialplan-tag Enters voice register dialplan configuration mode to define
a dial plan for SIP phones.
Example:
Router(config)# voice register dialplan 1
Example:
Router(config-register-dialplan)# exit
Step 7 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 4 to be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 8 dialplan dialplan-tag Assigns a dial plan to a SIP phone.
• dialplan-tag—Number that identifies the dial plan to
Example: use for this SIP phone. This is the number that was used
Router(config-register-pool)# dialplan 1 with the voice register dialplan command in Step 3.
Range: 1 to 24.
Step 9 end Exits to privileged EXEC mode.
Example:
Router(config-register-global)# end
Examples
The following example shows the configuration for dial plan 1 which is assigned to SIP phone 1.
voice register dialplan 1
type 7940-7960-others
pattern 1 2... timeout 10 user ip
pattern 2 1234 user ip button 4
pattern 3 65...
pattern 4 1...!
!
voice register pool 1
id mac 0016.9DEF.1A70
type 7961GE
number 1 dn 1
number 2 dn 2
dialplan 1
dtmf-relay rtp-nte
codec g711ulaw
What to Do Next
If you are done modifying parameters for SIP phones, you must generate a new configuration profile and
restart the phones. See “Generating Configuration Files for Phones” on page 245.
This command displays the configuration information for a specific SIP dial plan.
Router# show voice register dialplan 1
Dialplan Tag 1
Config:
Type is 7940-7960-others
Pattern 1 is 2..., timeout is 10, user option is ip, button is default
Pattern 2 is 1234, timeout is 0, user option is ip, button is 4
Pattern 3 is 65..., timeout is 0, user option is phone, button is default
Pattern 4 is 1..., timeout is 0, user option is phone, button is default
Pool Tag 29
Config:
Mac address is 0012.7F54.EDC6
Number list 1 : DN 29
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
keep-conference is enabled
dialplan tag is 1
kpml signal is enabled
service-control mechanism is not supported
.
.
.
Step 3 show voice register template tag
This command displays the dial plan assigned to a specific template.
Router# show voice register template 3
Temp Tag 3
Config:
Attended Transfer is disabled
Blind Transfer is enabled
Semi-attended Transfer is enabled
Conference is enabled
Caller-ID block is disabled
DnD control is enabled
Anonymous call block is disabled
Voicemail is 62000, timeout 15
Dialplan Tag is 1
Transport type is tcp
Prerequisites
• Cisco Unified CME 4.1 or a later version.
Restrictions
• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
• A dial plan assigned to a phone has priority over KPML.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. digit collect kpml
5. end
6. show voice register dial-peer
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 4 to be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 digit collect kpml Enables KPML digit collection for the SIP phone.
Note This command is enabled by default for supported
Example: phones in Cisco Unified CME.
Router(config-register-pool)# digit collect
kpml
Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-register-pool)# end
Step 6 show voice register dial-peers Displays details of all dynamically created VoIP dial peers
associated with the Cisco Unified CME SIP register
including the defined digit collection method.
Example:
Router# show voice register dial-peers
What to Do Next
If you are done modifying parameters for SIP phones, you must generate a new configuration profile and
restart the phones. See “Generating Configuration Files for Phones” on page 245.
Prerequisites
• Cisco Unified CME 4.1 or a later version.
• SIP phone to which configuration is to be applied must be already configured. For configuration
information, see the “SIP: Assigning Directory Numbers to Phones” section on page 163.
Restrictions
• TCP is not supported as a session-transport protocol for the Cisco Unified IP Phone 7905, 7912,
7940, or 7960. If TCP is assigned to an unsupported phone using this command, calls to that phone
will not complete successfully. The phone can originate calls, but it uses UDP, although TCP has
been assigned.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. session-transport {tcp | udp}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
What to Do Next
• If you want to disable SIP Proxy registration for an individual directory number, see the “SIP:
Disabling SIP Proxy Registration for a Directory Number” section on page 172.
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones”
on page 250
Prerequisites
• Cisco Unified CME 3.4 or a later version.
• Bulk registration is configured at system level. For configuration information, see “Configuring
Bulk Registration” on page 123.
Restrictions
• Phone numbers that are registered under voice register dn must belong to a SIP phone that is itself
registered in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. no-reg
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1
Step 4 number number Defines a valid number for a directory number to be
assigned to a SIP phone in Cisco Unified CME.
Example:
Router(config-register-dn)# number 4085550152
Step 5 no-reg Causes directory number being configured to not register
with an external proxy server.
Example:
Router(config-register-dn)# no-reg
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones”
on page 250
Configuring Codec for Local Calling Between SIP and SCCP Phones
To designate a codec for individual phones to ensure connectivity between SIP and SCCP phones
connected to the same Cisco Unified CME router, perform the following steps for each SIP or SCCP
phone.
Note If codec values for the dial peers of an internal connection do not match, the call fails.
Prerequisites
• Cisco Unified CME 3.4 or a later version.
• Cisco Unified IP phone to which codec is to be applied must be already configured. For
configuration information for SIP phones, see the “SIP: Assigning Directory Numbers to Phones”
section on page 163. For configuration information for SCCP phones, see the “SCCP: Assigning
Directory Numbers to Phones” section on page 160.
Restrictions
• Required only if you have SIP and SCCP phones connected to the same Cisco Unified CME router.
• Modify the configuration for either SIP or SCCP phones to ensure that the codec for all phones
match. Do not modify the configuration for SIP and SCCP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone ephone-tag
or
voice register pool-tag
4. codec codec-type
5. end
DETAILED STEPS
Example:
Router# configure terminal
What to Do Next
• If you want to select the session-transport protocol for a SIP phone, see the “SIP: Selecting
Session-Transport Protocol for a Phone” section on page 171.
• If you are finished configuring SIP phones to make basic calls using, you are ready to generate
configuration files for the phones to be connected. See “SIP: Generating Configuration Profiles for
SIP Phones” on page 250.
• If you are finished configuring SCCP phones to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.
Restrictions
• Do not configure directory numbers for a key system for dual-line mode because this does not
conform to the key system one-call-per-line button usage model for which the phone is designed.
• Provisioning support for the Cisco Unified IP Phone 7931 is available only in Cisco Unified CME
4.0(2) and later versions.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number [secondary number] [no-reg [both | primary]]
5. preference preference-order
6. no huntstop
or
huntstop
7. mwi-type {visual | audio | both}
8. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-ephone-dn)# number 101
Step 5 preference preference-order Sets dial-peer preference order for a directory number
associated with a Cisco Unified IP phone.
Example:
• Default: 0.
Router(config-ephone-dn)# preference 1
• Increment the preference order for all subsequent
instances within a set of ephone dns with the same
number to be associated with a key system phone.
That is, the first instance of the directory number is
preference 0 by default and you must specify 1 for
the second instance of the same number, 2 for the
next, and so on. This allows you to create multiple
buttons with the same number on an IP phone.
• Required to support call waiting and call transfer on
a key system phone.
Step 6 no huntstop Explicitly enables call hunting behavior for a directory
number.
Example: • Configure no huntstop for all instances, except the
Router(config-ephone-dn)# no huntstop final instance, within a set of ephone dns with the
same number to be associated with a key system
or phone.
huntstop
• Required to allow call hunting across multiple line
buttons with the same number on an IP phone.
Example: or
Router(config-ephone-dn)# huntstop
Disables call hunting behavior for a directory number.
• Configure the huntstop command for the final
instance within a set of ephone dns with the same
number to be associated with a key system phone.
• Required to limit the call hunting to a set of multiple
line buttons with the same number on an IP phone.
Example:
Router(config-ephone-dn)# end
Examples
The following example shows the configuration for six instances of directory number 101, assigned to
the first six buttons of an IP phone.
ephone-dn 10
number 101
no huntstop
ephone-dn 11
number 101
preference 1
no huntstop
ephone-dn 12
number 101
preference 2
no huntstop
ephone-dn 13
number 101
preference 3
no huntstop
ephone-dn 14
number 101
preference 4
no huntstop
ephone-dn 15
number 101
preference 5
ephone 1
mac-address 0001.2345.6789
type 7931
button 1:10 2:11 3:12 4:13 5:14 6:15
Prerequisites
• FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection
must be configured; for example:
voice-port 1/0/0
connection plar-opx 801 <<----Private number
Restrictions
• A directory number with a trunk line cannot be configured for call forward, busy, or no answer.
• Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed
on IP phones.
• Numbers entered after trunk line is seized will not appear in call history or call detail records
(CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.
• FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and
NewCall soft keys.
• FXO trunk lines do not support conference initiator dropoff.
• FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk
line before pressing the Redial button.
• FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference
an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP
phone connected. The conference initiator is unable to participate in the conference, but can place
calls on other lines.
• FXO trunk lines do not support bulk speed dial.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds] monitor-port port
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode to create a
directory number.
Example: • Configure this command in the default single line
Router(config)# ephone-dn 51 mode, without the dual-line keyword, when
configuring a simple key system trunk line.
Step 4 number number [secondary number] [no-reg [both | Configures a valid phone or extension number for this
primary]] directory number.
Example:
Router(config-ephone-dn)# number 801
Example:
Router(config-ephone-dn)# end
Examples
The following example shows the configuration for six instances of directory number 101, assigned to
the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10.
ephone-dn 10
number 101
no huntstop
ephone-dn 11
number 101
preference 1
no huntstop
ephone-dn 12
number 101
preference 2
no huntstop
ephone-dn 13
number 101
preference 3
no huntstop
ephone-dn 14
number 101
preference 4
no huntstop
ephone-dn 15
number 101
preference 5
ephone-dn 51
number 801
trunk 811 monitor-port 1/0/0
ephone-dn 52
number 802
trunk 812 monitor-port 1/0/1
ephone-dn 53
number 803
trunk 813 monitor-port 1/0/2
ephone-dn 54
number 804
trunk 814 monitor-port 1/0/3
ephone 1
mac-address 0001.2345.6789
type 7931
button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54
voice-port 1/0/0
connection plar opx 801
voice-port 1/0/1
connection plar opx 802
voice-port 1/0/2
connection plar opx 803
voice-port 1/0/3
connection plar opx 804
What to Do Next
You are ready to configure each individual phone and assign button numbers, line characteristics, and
directory numbers to buttons on the phone. See the “SCCP: Configuring Individual IP Phones for Key
System” section on page 187.
Prerequisites
• FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection
must be configured; for example:
voice-port 1/0/0
connection plar-opx 801 <<----Private number
Restrictions
• An ephone-dn with a trunk line cannot be configured for call forward, busy, or no answer.
• Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed
on IP phones.
• Numbers entered after trunk line is seized will not appear in call history or call detail records
(CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.
• FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and
NewCall soft keys.
• FXO trunk lines do not support conference initiator dropoff.
• FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk
line before pressing the Redial button.
• FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference
an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP
phone connected. The conference initiator is unable to participate in the conference, but can place
calls on other lines.
• FXO trunk lines do not support bulk speed dial.
• FXO port monitoring has the following restrictions:
– Not supported before Cisco Unified CME 4.0.
– Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports.
FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.
– Not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
– T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot
into a ds0-group).
• Transfer recall and transfer-to button optimization is supported on dual-line directory numbers only
in Cisco Unified CME 4.0 and later.
• Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on
hold, or call pickup at alert.
• Transfer recall is not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag dual-line
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds] [transfer-timeout seconds] [monitor-port port]
6. huntstop [channel]
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag dual-line Enters ephone-dn configuration mode for the purposes of
creating and configuring a telephone or extension
number.
Example:
Router(config)# ephone-dn 51 dual-line • dual-line—Required when configuring an advanced
key system phone trunk line. Dual-line mode
provides a second call channel for the directory
number on which to place an outbound consultation
call during the call transfer attempt. This also allows
the phone to remain part of the call in order to
monitor the progress of the transfer attempt and if
the transfer is not answered, to pull the call back to
the phone on the original PSTN line button.
Step 4 number number [secondary number] [no-reg [both | Configures a valid telephone number or extension
primary]] number for this directory number.
Example:
Router(config-ephone-dn)# number 801
Example:
Router(config-ephone-dn)# end
Examples
The following example shows the configuration for six instances of directory number 101, assigned to
the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10.
These four PSTN line appearances are configured as dual lines to provide a second call channel on which
to place an outbound consultation call during a call transfer attempt. This configuration allows the phone
to remain part of the call in order to monitor the progress of the transfer attempt, and if the transfer is
not answered, to pull the call back to the phone on the original PSTN line button.
ephone-dn 10
number 101
no huntstop
ephone-dn 11
number 101
preference 1
no huntstop
ephone-dn 12
number 101
preference 2
no huntstop
ephone-dn 13
number 101
preference 3
no huntstop
ephone-dn 14
number 101
preference 4
no huntstop
ephone-dn 15
number 101
preference 5
ephone-dn 51 dual-line
number 801
trunk 811 transfer-timeout 30 monitor-port 1/0/0
huntstop channel
ephone-dn 52 dual-line
number 802
trunk 812 transfer-timeout 30 monitor-port 1/0/1
huntstop channel
ephone-dn 53 dual-line
number 803
trunk 813 transfer-timeout 30 monitor-port 1/0/2
huntstop channel
ephone-dn 54 dual-line
number 804
trunk 814 transfer-timeout 30 monitor-port 1/0/3
huntstop channel
ephone 1
mac-address 0001.2345.6789
type 7931
button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54
voice-port 1/0/0
connection plar opx 801
voice-port 1/0/1
connection plar opx 802
voice-port 1/0/2
connection plar opx 803
voice-port 1/0/3
connection plar opx 804
Restrictions
• Provisioning for Cisco Unified IP Phone 7931G is available only in Cisco Unified CME 4.0(2) and
later versions.
• Cisco Unified IP Phone 7931G can support only one call waiting overlaid per directory number.
• Cisco Unified IP Phone 7931G cannot support overlays that contain directory numbers configured
for dual-line mode.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address [mac-address]
5. type phone-type
6. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]
7. mwi-line line-number
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 1
Step 4 mac-address [mac-address] Specifies the MAC address of the IP phone that is being
configured.
Example:
Router(config-ephone)# mac-address 0001.2345.6789
Example:
Router(config-ephone)# type 7931
Step 6 button button-number{separator}dn-tag Associates a button number and line characteristics with
[,dn-tag...] an ephone-dn. Maximum number of buttons is
[button-number{x}overlay-button-number]
[button-number...]
determined by phone type.
Tip The line button layout for the Cisco Unified IP
Phone 7931G is a bottom-up array. Button 1 is at
Example:
the bottom right of the array and button 24 is at
Router(config-ephone)# button 1:11 2:12 3:13 4:14
5:15 6:16 7:51 8:52 9:53 10:54 the top left of the array.
Step 7 mwi-line line-number Selects a phone line to receive MWI treatment; when a
message is waiting for the selected line, the message
waiting indicator is activated.
Example:
Router(config-ephone)# mwi-line 3 • line-number—Range: 1 to 34. Default: 1.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones”
on page 247.
Restrictions
For a Cisco ATA that is registered to a Cisco Unified CME system to participate in fax calls, it must have
its ConnectMode parameter set to use the same RTP payload type as the Cisco voice gateway that is
performing the fax pass-through. Cisco voice gateways use standard payload type 0/8, which is selected
on Cisco ATAs by setting bit 2 of the ConnectMode parameter to 1. For more information, see the
“Parameters and Defaults” chapter in the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor
Administrator's Guide for SCCP (version 3.0).
SUMMARY STEPS
DETAILED STEPS
Step 1 Install the Cisco ATA. See the “Installing the Cisco ATA” chapter in the in Cisco ATA 186 and
Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SCCP (version 3.0).
Step 2 Configure the Cisco ATA. See the “Configuring the Cisco ATA for SCCP” chapter in the Cisco ATA 186
and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SCCP (version 3.0).
Step 3 Upgrade to the latest Cisco ATA image. If you are using either the v2.14 or v2.14ms Cisco ATA 186
image based on the 2.14 020315a build for H.323/SIP or the 2.14 020415a build for MGCP or SCCP,
you must upgrade to the latest version to install a security patch. This patch fixes a security hole in the
Cisco ATA Web server that allows users to bypass the user interface password.
For information about upgrading firmware, see “Installing and Upgrading Cisco Unified CME
Software” on page 69. Alternatively, you can use a manual method, as described in the “Upgrading the
Cisco ATA Signaling Image” chapter of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone
Adaptor Administrator’s Guide for SCCP (version 3.0).
Step 4 Configure the Cisco ATA to set the following parameters:
– DHCP parameter to 1 (enabled).
– TFTP parameter to 1 (enabled).
– TFTPURL parameter to the IP address of the router running Cisco Unified CME.
– SID0 parameter to a period (.) or the MAC address of the Cisco ATA (to enable the first port).
– SID1 parameter to a period (.) or a modified version the Cisco ATA’s MAC address, with the
first two hexadecimal numbers removed and 01 appended to the end, if you want to use the
second port. For example, if the MAC address of the Cisco ATA is 00012D01073D, set SID1 to
012D01073D01.
– Nprintf parameter to the IP address and port number of the host to which all Cisco ATA debug
messages are sent. The port number is usually set to 9001.
– To prevent tampering and unauthorized access to the Cisco ATA 186, you can disable the
web-based configuration. However, if you disable the web configuration page, you must use
either a TFTP server or the voice configuration menu to configure the Cisco ATA 186.
Step 5 Configure analog phones that use a Cisco ATA in the same way as a Cisco Unified IP phone. In the type
command, use the ata keyword. For information on how to provision phones, see the “SCCP: Creating
Directory Numbers” section on page 158.
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones”
on page 247 and “SIP: Generating Configuration Profiles for SIP Phones” on page 250.
Using Call Pickup and Group Call Pickup with Cisco ATA
Most of the procedures for using Cisco ATAs with Cisco Unified CME are the same as those for using
Cisco ATAs with Cisco Unified Communications Manager, as described in the “How to Use Pre-Call
and Mid-Call Services” chapter of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor
Administrator’s Guide for SCCP (version 3.0). However, the call pickup and group call pickup
procedures are different when using Cisco ATAs with Cisco Unified CME, as described below:
Call Pickup
When using Cisco ATAs with Cisco Unified CME:
• To pickup the last parked call, press **3*.
• To pickup a call on a specific extension, press **3 and enter the extension number.
• To pickup a call from a park slot, press **3 and enter the park slot number.
Note If there is only one pickup group, you do not need to enter the group ID after the **4 to pickup a call.
Prerequisites
• Cisco CME 3.2.2 or a later version for analog FXS ports on the Cisco VG 224 Voice Gateway.
• Cisco Unified CME 4.0 or a later version for analog FXS ports on the Cisco 2800 Series or
Cisco 3800 Series Integrated Services Routers.
Restrictions
• FXS ports on Cisco VG 248 Analog Phone Gateways are not supported by Cisco Unified CME.
• You must set the transfer-system command to full-blind or full-consult to enable call transfer on
analog endpoints.
• You must set the timeouts ringing command to infinity (default) on the analog ports to prevent this
timeout from expiring before the ringing no-answer timeout that is configured on
Cisco Unified CME with the timeouts ringing command in telephony-service mode.
Note In Cisco IOS Release 12.4(11)T and later the default value of the timeouts ringing
command is set to infinity for all SCCP-controlled analog ports. In releases earlier than
Cisco IOS Release 12.4(11)T, the default is 180 seconds.
SUMMARY STEPS
DETAILED STEPS
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.
Prerequisites
• The WAN link supporting remote teleworker phones should be configured with a Call Admission
Control (CAC) or Resource Reservation Protocol (RSVP) solution to prevent the oversubscription
of bandwidth, which can degrade the quality of all voice calls.
• If DSP farms will be used for transcoding, you must configure them separately. See “Configuring
Transcoding Resources” on page 303.
• A SCCP phone to be enabled as a remote phone is configured in Cisco Unified CME. For
configuration information, see the “SCCP: Creating Directory Numbers” section on page 158
Restrictions
• Because Cisco Unified CME is not designed for centralized call processing, remote phones are
supported only for fixed teleworker applications, such as working from a home office.
• Cisco Unified CME does not support CAC for remote SCCP phones, so voice quality can degrade
if a WAN link is oversubscribed. High-bandwidth data applications used over a WAN can cause
degradation of voice quality for remote IP phones.
• Cisco Unified CME does not support Emergency 911 (E911) calls from remote IP phones.
Teleworkers using remote phones connected to Cisco Unified CME over a WAN should be advised
not to use these phones for E911 emergency services because the local public safety answering point
(PSAP) will not be able to obtain valid calling-party information from them.
We recommend that you make all remote phone users aware of this issue. One way is to place a label
on all remote teleworker phones that reminds users not to place 911 emergency calls on remote IP
phones. Remote workers should place any emergency calls through locally configured hotel, office,
or home phones (normal land-line phones) whenever possible. Inform remote workers that if they
must use remote IP phones for emergency calls, they should be prepared to provide specific location
information to the answering PSAP personnel, including street address, city, state, and country.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mtp
5. codec {g711ulaw | g729r8 [dspfarm-assist]}
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks.
Router(config)# ephone 36
Step 4 mtp Sends media packets to the Cisco Unified CME router.
Example:
Router(config-ephone)# mtp
Step 5 codec {g711ulaw | g729r8 [dspfarm-assist]} (Optional) Selects a preferred codec for setting up calls.
• g711ulaw—G.711 mu-law codec (default).
Example: • g729r8—G.729r8 codec.
Router(config-ephone)# codec g729r8
dspfarm-assist • dspfarm-assist—Attempts to use DSP-farm resources
for transcoding the segment between the phone and the
Cisco Unified CME router if G.711 is negotiated for
the call.
Note The dspfarm-assist keyword is ignored if the SCCP
endpoint type is ATA, VG224, or VG248.
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.
Prerequisites
• Cisco Unified CME 4.0 or a later version
• Cisco IP Communicator 2.0 or a later version
• IP address of the Cisco Unified CME TFTP server
• (Optional) Headsets with microphones for users
DETAILED STEPS
Step 1 Download the latest version of the Cisco IP Communicator software and install it on your PC.
The download website is at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.
Step 2 (Optional) Attach a headset with microphone to your PC.
Step 3 Start the Cisco IP Communicator application.
Step 4 Define the IP address of the Cisco Unified CME TFTP server.
a. Open the Network > User Preferences window.
b. Enter the IP address of the Cisco Unified CME TFTP server.
Step 5 Wait for the Cisco IP Communicator application to connect to Cisco Unified CME and register.
Step 6 Configure the extension numbers and line buttons for the Cisco IP Communicator.
Use the normal phone provisioning commands described in the “SCCP: Creating Directory Numbers”
section on page 158. In the type command, use the CIPC keyword to identify this phone as a
Cisco IP Communicator.
version 12.4
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CME40
!
boot-start-marker
boot-end-marker
!
logging buffered 2000000 debugging
!
no aaa new-model
!
resource policy
!
clock timezone PST -8
clock summer-time PDT recurring
no network-clock-participate slot 2
voice-card 0
no dspfarm
dsp services dspfarm
!
voice-card 2
dspfarm
!
no ip source-route
ip cef
!
!
!
ip domain name cisco.com
ip multicast-routing
!
!
ftp-server enable
ftp-server topdir flash:
isdn switch-type primary-5ess
!
!
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
no supplementary-service h450.2
no supplementary-service h450.3
h323
call start slow
!
!
!
controller T1 2/0/0
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 2/0/1
framing esf
linecode b8zs
!
!
interface GigabitEthernet0/0
ip address 192.168.1.1 255.255.255.0
ip pim dense-mode
duplex auto
speed auto
media-type rj45
negotiation auto
!
interface Service-Engine1/0
ip unnumbered GigabitEthernet0/0
service-module ip address 192.168.1.2 255.255.255.0
service-module ip default-gateway 192.168.1.1
!
interface Serial2/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn incoming-voice voice
isdn map address ^.* plan unknown type international
no cdp enable
!
!
ip route 0.0.0.0 0.0.0.0 192.168.1.254
ip route 192.168.1.2 255.255.255.255 Service-Engine1/0
ip route 192.168.2.253 255.255.255.255 10.2.0.1
ip route 192.168.3.254 255.255.255.255 10.2.0.1
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
!
!
tftp-server flash:P00307020300.loads
tftp-server flash:P00307020300.sb2
tftp-server flash:P00307020300.sbn
!
control-plane
!
!
!
voice-port 2/0/0:23
!
!
!
sccp local GigabitEthernet0/0
sccp ccm 192.168.1.1 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register MTP0013c49a0cd0
keepalive retries 5
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec gsmfr
codec g729r8
maximum sessions 90
associate application SCCP
!
!
dial-peer voice 9000 voip
mailbox-selection last-redirect-num
destination-pattern 78..
session protocol sipv2
session target ipv4:192.168.1.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2 pots
incoming called-number .
direct-inward-dial
port 2/0/0:23
forward-digits all
!
dial-peer voice 1 pots
destination-pattern 9[2-9]......
port 2/0/0:23
forward-digits 8
!
dial-peer voice 3 pots
destination-pattern 91[2-9]..[2-9]......
port 2/0/0:23
forward-digits 12!
!
gateway
timer receive-rtp 1200
!
!
telephony-service
load 7960-7940 P00307020300
max-ephones 100
max-dn 300
ip source-address 192.168.1.1 port 2000
system message CCME 4.0
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 MTP0013c49a0cd0
voicemail 7800
max-conferences 24 gain -6
call-forward pattern .T
moh music-on-hold.au
multicast moh 239.1.1.1 port 2000
web admin system name admin password sjdfg
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn-template 1
!
!
ephone-template 1
keep-conference endcall local-only
codec g729r8 dspfarm-assist
!
!
ephone-template 2
!
!
ephone-dn 1
number 6001
call-forward busy 7800
call-forward noan 7800 timeout 10
!
!
ephone-dn 2
number 6002
!
!
end
external-ring bellcore-dr3
voice register dn 1
number 2300
mwi
voice register dn 2
number 2200
call-forward b2bua all 1000
call-forward b2bua mailbox 2200
mwi
voice register dn 3
number 2201
after-hour exempt
voice register dn 4
number 2100
call-forward b2bua busy 2000
mwi
voice register dn 5
number 2101
mwi
voice register dn 76
number 2525
call-forward b2bua unreachable 2300
mwi
!
voice register template 1
!
voice register template 2
no conference enable
voicemail 7788 timeout 5
!
dtmf-relay rtp-nte
speed-dial 3 2001
speed-dial 4 2201
!
voice register pool 3
id mac 0030.94C3.053E
type 7960
number 1 dn 3
number 3 dn 3
template 2
!
voice register pool 5
id mac 0012.019B.3FD8
type ATA
number 1 dn 5
preference 1
dtmf-relay rtp-nte
codec g711alaw
template 2
dtmf-relay-rtp-nte
call-forward b2bua all 7778
voice-port 1/0/0
voice-port 1/0/1
sip-ua
mwi-server ipv4:1.15.6.200 expires 3600 port 5060 transport udp
telephony-service
load 7960-7940 P0S3-07-2-00
max-ephones 24
max-dn 96
ip source-address 10.15.6.112 port 2000
create cnf-files version-stamp Aug 24 2004 00:00:00
max-conferences 8
after-hours block pattern 1 1...
after-hours day Mon 17:00 07:00
voice register dn 1
number 4085550101
no-reg
sip-ua
registrar ipv4:1.5.49.240
!
!
ephone-dn 10 dual-line
number 4443 secondary 9191114443
pickup-group 5
description vg224-1/3
name tommy
!
ephone 7
mac-address C863.9018.0402
speed-dial 1 4445
speed-dial 2 4445
speed-dial 3 4442
speed-dial 4 4441
speed-dial 5 6666
speed-dial 6 1111
speed-dial 7 1112
speed-dial 8 9191114441
speed-dial 9 9191114442
speed-dial 10 9191114442
type anl
button 1:10
!
Where to Go Next
To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button Layout
for a Cisco Unified IP Phone 7931G” on page 897 .
After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected to your router. See “Generating Configuration Files
for Phones” on page 245.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 8 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the Extension Assigner feature in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Prerequisites for Extension Assigner, page 213
• Restrictions for Extension Assigner, page 214
• Information About Extension Assigner, page 214
• How to Configure Extension Assigner, page 219
• Configuration Examples for Extension Assigner, page 239
• Additional References, page 242
• Feature Information for Extension Assigner, page 243
If you decide to use the ephone tag, it will require less configuration. However, the installation technician
will enter an arbitrary tag number instead of the actual extension number when configuring a phone. This
restriction is because the number of ephone tags that you can configure is limited by your license. For
example, if you use the ephone tag and you have a 100-user license, the installation technician cannot
enter 9001 for the tag because you can configure only ephone 1 to ephone 100.
Note that each ephone entry that you configure must also include a temporary MAC address. As shown
in the above example, this address should begin with 02EA.EAEA and can end with any unique number.
We strongly recommend that you can configure this unique number to match the ephone tag.
You do not have to configure any ephone entries for the extension number that are randomly assigned.
The autoassign feature automatically creates an ephone entry for each new phone when it registers. The
autoassign feature then automatically assigns an ephone-dn entry if there is an available ephone-dn that
has one of the tag numbers specified by the auto assign command. The resulting ephone configurations
have the actual MAC address of the phone and a button with the first available ephone-dn designated for
the autoassign feature.
As shown in the following example, you configure at least one ephone-dn for a temporary extension and
specify which ephone-dns the autoassign feature will assign to the temporary ephone entries:
telephony-service
auto assign 101 to 105
ephone-dn 101
number 0001
When the installation technician assigns an extension number to a phone, the temporary MAC address
is replaced by the actual MAC address and the ephone entry created by the autoregister feature is deleted.
The number of ephone-dns that you configure for the autoassign feature determines how many phones
you can plug in at one time and get an automatically assigned extension. If you define four ephone-dns
for autoassign and you plug in five phones, one phone will not get a temporary extension number until
you assign an extension to one of the other four phones and reset the fifth phone. You are permitted to
set the max-ephone value higher than the number of purchased Cisco Unified CME phone seat licenses
for the purpose of enrolling licensed phones using extension assigner.
In addition to configuring one ephone-dn for each temporary extension number that is assigned
automatically, you also must configure an ephone-dn entry for each extension number that is assigned
by the installation technician.
Therefore, to complete the configuration, as shown in the following example, you must:
• Specify whether to use the ephone or the provision-tag number to reference the extension
number to assign to the phone. Set this when the feature is enabled with the new
extension-assigner tag-type command provided with this feature.
• Configure an ephone-dn for each temporary extension number that is assigned automatically.
• Configure an ephone-dn for each extension number that you want the installation technician to
assign to a phone.
• Configure an ephone with a temporary MAC address for each phone that is assigned an extension
number by the installation technician. Optionally, this ephone definition can include the new
provision-tag. For more information, see the “Configuring Ephones with Temporary MAC
Addresses” section on page 231.
telephony-service
extension-assigner tag-type provision-tag
auto assign 101 to 105
ephone-dn 1 dual-line
number 6001
ephone-dn 101
number 0001
label Temp-Line-not assigned yet
ephone 1
provision-tag 6001
mac-address 02EA.EAEA.0001
button 1:1
Because you must configure two ephone-dns for each extension number that you want to assign, you may
exceed your max-dn setting. You are permitted to set the max-dn value higher than the number allowed
by your license for the purpose of enrolling licensed phones using extension assigner.
Assuming that your max-dn setting is set high enough, your max-ephone setting determines how many
phones you should plug in at one time. For example, if your max-ephone setting is ten more than the
number of phones to which you want to assign extension numbers, the you can plug in ten phones at a
time. If you plug in eleven phones, one phone will not register or get a temporary extension number until
you assign an extension to one of the first ten phones and reset the eleventh phone.
After you have configured your ephone and ephone-dn entries, you can complete your router
configuration by optionally configuring the router to automatically save your configuration. If the router
configuration is not saved, any extension assignments made by the installation technician will be lost
when the router is restarted. The alternative to this optional procedure is to have the installation
technician connect to the router and enter the write memory command to save the router configuration.
The final task of the system administrator is to document the information that the installation technician
needs to assign extension numbers to the new phones. You can also use this documentation as a guide
when you configure Cisco Unified CME to implement this feature. This information includes:
• How many phones the installation technician can plug in at one time
• Which extension number to dial to access the extension assigner application
• Whether the number is dialed automatically when a phone goes off hook
• What password to enter to access the application
• Which tag numbers to enter to assign en extension to each phone
Note Because this feature is implemented using a Tcl script and audio files, you must place the script and
associated audio prompt files in the correct directory. Do not edit this script; just configure
Cisco Unified CME to load the appropriate script.
The system administrator provides the installation technician with all of the information needed to
perform this procedure.
Determining Which Extension Numbers to Assign to the New Phones and Plan
Your Configuration
After you determine which extension number to assign to each phone, you must make the following
decisions:
• Which extension number must be dialed to access the extension assigner application.
• Whether the number is dialed automatically when a phone goes off hook.
• What password the installation technician must enter to access the extension assigner application.
• Whether to use the ephone or the provision-tag number to reference the extension number to assign
to the phone.
• How many temporary extension numbers to configure. This will determine how many temporary
ephone-dns and temporary MAC addresses to configure.
• What specific tag numbers to use to reference the extension number to assign to the phone.
SUMMARY STEPS
DETAILED STEPS
Example:
Router# enable
Step 4 archive tar /xtract source-url destination-url Uncompresses the files in the Cisco Unified CME extension
assigner app-cme-ea-2.0.0.0.tar or later archive file and
copies them to a location that is accessible by the
Example:
Router# archive tar /xtract
Cisco Unified CME router.
tftp://192.168.1.1/app-cme-ea-2.0.0.0.tar • source-url—URL of the source of the extension
flash:
assigner TAR file. Valid URLs can refer to TFTP or
HTTP servers or to flash memory.
• location—URL of the destination of the extension
assigner TAR file, including its Tcl script and audio
files. Valid URLs can refer to TFTP or HTTP servers or
to flash memory.
Note For extension assigner, you must configure the Tcl script to use English as the language.
Tip To change the password, you must remove the existing extension assigner service and create a new
service that defines a new password.
SUMMARY STEPS
1. enable
2. configure terminal
3. application
4. service service-name location
5. param ea-password password
6. paramspace english index number
7. paramspace english language en
8. paramspace english location location
9. paramspace english prefix en
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 application Enters application configuration mode to configure
packages and services.
Example:
Router(config)# application
Step 4 service service-name location Enters service parameter configuration mode to configure
parameters for the call-queue service.
Example: • service-name—Name of the extension assigner service.
Router(config-app)# service EA This arbitrary name is used to identify the service
tftp://10.1.1.100/app-cme-ea-2.0.0.0.tcl during configuration tasks.
• location—URL of the Tcl script for the extension
assigner service. Valid URLs can refer to TFTP or
HTTP servers or to flash memory.
Step 5 param ea-password password Sets the password that installation technicians enter to
access the extension assigner application.
Example: • password—Numerical password that installation
Router(config-app-param)# param ea-password technicians enter to access the extension assigner
1234 application. It can be 2 to 10 digits long.
Example:
Router(config-app-param)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
4. service service-name outbound
5. destination-pattern string
6. session target ipv4:destination-address
7. dtmf-relay h245-alphanumeric
8. codec g711ulaw
9. no vad
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.
• tag—Number used during configuration tasks to
Example: identify this dial peer.
Router(config)# dial-peer voice 5999 voip
Step 4 service service-name outbound Loads and configures the extension assigner application on
a dial peer.
Example: • service-name—Name that identifies the voice
Router(config-dial-peer)# service EA out-bound application. This is a user-defined name and does not
have to match the script name. In this case, the name
must match the name that you used to load the
extension assigner Tcl script in the “Configuring the
Tcl Script” section on page 221.
• outbound—Indicates that this is an outbound dial peer.
It is required for extension assigner.
Example:
Router(config-dial-peer)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. extension-assigner tag-type {ephone-tag | provision-tag}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds]
6. name name
7. exit
8. telephony-service
9. auto assign dn-tag to dn-tag
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 90 ephone-dn during configuration tasks.
• dual-line—(Optional) Enables an ephone-dn with one
voice port and two voice channels, which supports
features such as call waiting, call transfer, and
conferencing with a single ephone-dn.
Note We recommend that you use single-line mode for
your temporary extension numbers.
Step 4 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn
| primary]] instance.
• number—String of up to 16 digits that represents a
Example: telephone or extension number to be associated with
Router(config-ephone-dn)# number 9000 this ephone-dn.
• secondary—(Optional) Allows you to associate a
second telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should
not register with the H.323 gatekeeper. Unless you
specify one of the optional keywords (both or
primary) after the no-reg keyword, only the
secondary number is not registered.
Step 5 trunk digit-string [timeout seconds] (Optional) Configures the extension number to
automatically dial the extension assigner application.
Example: • digit-string—The number of the extension assigner
Router(config-ephone-dn)# trunk 9000 application. This number must match the number that
you configured in “Specifying the Extension That
Installation Technicians Call to Assign Extension
Numbers” section on page 224
• timeout seconds—(Optional) Interdigit timeout
between dialed digits, in seconds. Range is 3 to 30.
Default is 3.
Example:
Router(config-ephone-dn)# exit
Step 8 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 9 auto assign dn-tag to dn-tag Automatically assigns ephone-dn tags to Cisco Unified IP
phones as they register for service with a
Cisco Unified CME router. The ephone-dn tags that you
Example:
Router(config-telephony)# auto assign 90 to 99
specify in this command must match the tags that you
configured earlier in this procedure.
Step 10 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. name name
6. end
Note Repeat steps 3 to 5 for each extension number that you want to assign.
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 20 ephone-dn during configuration tasks.
• dual-line—(Optional) Enables an ephone-dn with one
voice port and two voice channels, which supports
features such as call waiting, call transfer, and
conferencing with a single ephone-dn.
Note To change an ephone-dn from dual-line to
single-line mode or the reverse, you must first
delete the ephone-dn and then recreate it.
Step 4 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn
| primary]] instance.
• number—String of up to 16 digits that represents a
Example: telephone or extension number to be associated with
Router(config-ephone-dn)# number 20 this ephone-dn.
• secondary—(Optional) Allows you to associate a
second telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should
not register with the H.323 gatekeeper. Unless you
specify one of the optional keywords (both or
primary) after the no-reg keyword, only the
secondary number is not registered.
Step 5 name name (Optional) Associates a name with this ephone-dn instance.
This name is used for caller-ID displays and in the local
directory listings.
Example:
Router(config-ephone-dn)# name hardware • You must follow the name order that is specified in the
directory command in telephony-service configuration
mode (either first-name-first or last-name-first).
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
Note If you want to use Cisco VG224 analog voice gateways with extension assigner, you need a minimum of
24 temporary ephones available for each gateway because they will attempt to temporary register all 24
of their ports as ephones.
You are permitted to set the max-ephone value higher than the number of purchased CME phone seat
licenses is for the purpose of enrolling licensed phones using extension assigner.
The readme file provided with this feature contains some sample entries for this procedure that you can
edit to fit your needs.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. provision-tag number
5. mac-address 02EA.EAEA.number
6. type phone-type [addon 1 module-type [2 module-type]]
7. button
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks. The maximum
Router(config)# ephone 20 number of ephones is version and platform-specific.
Type ? to display range.
If you use the ephone-tag keyword with the
extension-assigner tag-type command, this tag is used
to reference the extension number and must match the
number that the installation technician enters when
assigning an extension.
Step 4 provision-tag number (Optional) Specifies a unique sequence number that is used
by the extension assigner application only if you use the
provision-tag keyword with the extension-assigner
Example:
Router(config-ephone)# provision-tag 20
tag-type command.
• number—Unique sequence number that identifies
which ephone configuration and extension numbers to
assign to a phone. This number must match the number
that the installation technician enters when assigning an
extension.
Step 5 mac-address 02EA.EAEA.number Specifies a temporary MAC address number for this
ephone. For the extension assigner, this MAC address
should begin with 02EA.EAEA.
Example:
Router(config-ephone)# mac-address • number—We strongly recommends that you make this
02EA.EAEA.0020 number the same as the ephone number.
Example:
Router(config-ephone)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. kron policy-list list-name
4. cli write
5. exit
6. kron occurrence occurrence-name [user username] in [[numdays:]numhours:]nummin {oneshot |
recurring}
7. policy-list list-name
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 kron policy-list list-name Specifies a name for a new or existing Command Scheduler
policy list and enters kron-policy configuration mode.
Example: • If the value of the list-name argument is new, a new
Router(config)# kron policy-list save-config policy list structure is created.
• If the value of the list-name argument exists, the
existing policy list structure is accessed. No editor
function is available, and the policy list is run in the
order in which it was configured.
• Specifies a Command Scheduler policy list.
Step 4 cli write Specifies the fully-qualified EXEC command and
associated syntax to be added as an entry in the specified
Command Scheduler policy list. In this case, we want to
Example:
Router(config-kron-policy)# cli write
save the router configuration, so the command is write.
Example:
Router(config-kron-policy)# exit
Step 6 kron occurrence occurrence-name [user username] Specify schedule parameters for a Command Scheduler
[[in numdays:]numhours:]nummin {oneshot | occurrence and enters kron-occurrence configuration mode.
recurring}
We recommend that you configure your router to save your
configuration every 30 minutes.
Example: • occurrence-name—Specifies the name of the
Router(config)# kron occurrence backup in 30
occurrence. Length of occurrence-name is from 1 to 31
recurring
characters. If the occurrence-name is new, an
occurrence structure will be created. If the
occurrence-name is not new, the existing occurrence
will be edited.
• user—(Optional) Used to identify a particular user.
• username—Name of user.
• in—Identifies that the occurrence is to run after a
specified time interval. The timer starts when the
occurrence is configured.
• numdays:—(Optional) Number of days. If used, add a
colon after the number.
• numhours:—(Optional) Number of hours. If used, add
a colon after the number.
• nummin:—(Optional) Number of minutes.
• oneshot—Identifies that the occurrence is to run only
one time. After the occurrence has run, the
configuration is removed.
• recurring—Identifies that the occurrence is to run on a
recurring basis.
Step 7 policy-list list-name Specifies a Command Scheduler policy list.
Example:
Router(config-kron-occurrence)# policy-list
save-config
Step 8 end Returns to privileged EXEC mode.
Example:
Router(config-kron-occurrence)# end
Step 1 Get the information you need to use extension assigner from your system administrator. For a list of this
information, see the “Provide the Installation Technician with the Needed Information” section on
page 236.
Step 2 Dial the appropriate extension number to access the extension assigner system.
Step 3 Enter the password for the extension assigner and press #.
Step 4 Enter the ID number that represents this phone’s extension and press #.
Step 5 If the extension is not assigned to another phone, press 1 to confirm that you want to assign the extension
to your phone, then hang up. After the phone resets, the assignment is complete.
Step 6 If the extension is assigned to another phone that is idle:
a. Press 2 to confirm that you want to unassign the extension from the other phone.
b. Hang up.
c. Repeat this procedure beginning at Step 2.
Step 7 If the extension is assigned to another phone that is in use, either:
• Return to Step 5 to enter another extension number.
• Perform the procedures in the “Unassigning the Current Extension Number” section on page 237
and then repeat this procedure beginning at Step 2.
Step 1 Get the information you need to use extension assigner from your system administrator. For a list of this
information, see the “Provide the Installation Technician with the Needed Information” section on
page 236.
Step 2 Dial the appropriate extension number to access the extension assigner system.
Step 3 Enter the password for the extension assigner and press #.
Step 4 Enter the ID number that represents this phone’s extension and press #.
Step 5 When you enter the ID number for the extension that is currently assigned to this phone, you are
prompted to press 2 to confirm that you want to unassign the extension from the phone.
Step 6 Hang up.
Step 1 Get the information you need to use extension assigner from your system administrator. For a list of this
information, see the “Provide the Installation Technician with the Needed Information” section on
page 236.
Step 2 Dial the appropriate extension number to access the extension assigner system.
Step 3 Enter the password for the extension assigner and press #.
Step 4 Enter the ID number that represents this phone’s extension and press #.
Step 5 If the extension is not assigned to another phone, press 1 to confirm that you want to assign the extension
to your phone, then hang up. After the phone resets, the reassignment is complete.
Step 6 If the extension is assigned to another phone that is idle:
a. Press 2 to confirm that you want to unassign the extension from the other phone.
b. Hang up
c. Perform the procedure in the “Assigning New Extension Numbers” section on page 236.
Step 7 If the extension is assigned to another phone that is in use, either:
• Return to Step 5 to enter another extension number.
• Perform the procedures in the “Unassigning the Current Extension Number” section on page 237
and “Assigning New Extension Numbers” section on page 236.
Step 2 Use the debug voip application script command to display status messages produced by the server as
it runs the assigner application Tcl script.
Jun 20 23:17:45.795: //22//TCL :/tcl_PutsObjCmd: TCL: ***** >>> app-cme-ea-2.0.0.0.tcl <<<
*****
Jun 20 23:17:45.799: //22//TCL :/tcl_PutsObjCmd: TCL: ***** >>> Cisco CME Extension
Assigner Application <<< ****
Jun 20 23:17:45.799: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Enter password <<<
Jun 20 23:17:54.559: //22//TCL :/tcl_PutsObjCmd: >>> Collect Password Status = cd_005 <<<
Jun 20 23:17:54.563: //22//TCL :/tcl_PutsObjCmd: >>> INFO: Authentication Successful <<<
Jun 20 23:17:54.563: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Please enter the phone tag
number followed by the # key. Press * to re-enter the tag number <<<
Jun 20 23:17:59.839: //22//TCL :/tcl_PutsObjCmd: >>> Ephone TAG Digit Collect Status =
cd_005 <<<
Jun 20 23:17:59.843: //22//TCL :/tcl_PutsObjCmd: >>> INFO: Phone Query result = 1 <<<
Jun 20 23:17:59.843: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Ephone Tag 6 is available
<<<
Jun 20 23:17:59.843: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: To assign extension to Phone,
press 1 to confirm, 9 to cancel <<<
Jun 20 23:17:59.851: //22//TCL :/tcl_PutsObjCmd: >>> INFO: ephone 6 is available <<<
Jun 20 23:18:20.375: //22//TCL :/tcl_PutsObjCmd: >>> INFO: TAPS Status = cd_005 <<<
Jun 20 23:18:20.379: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Extension assignment is
successful <<<
Jun 20 23:18:20.379: //22//TCL :/tcl_PutsObjCmd: >>> INFO: Ephone extension is assigned
successfully <<<
Jun 20 23:18:28.975: //22//TCL :/tcl_PutsObjCmd: **** >>> TCL: Closing Cisco CM
Step 3 Use the debug ephone state command as described in the Cisco IOS Debug Command Reference.
version 12.4
no service password-encryption
!
hostname Test-Router
!
boot-start-marker
boot system flash:c2800nm-ipvoice-mz.2006-05-31.GOPED_DEV
boot-end-marker
!
enable password ww
!
no aaa new-model
!
resource policy
!
ip cef
no ip dhcp use vrf connected
!
ip dhcp pool pool21
network 172.21.0.0 255.255.0.0
default-router 172.21.200.200
option 150 ip 172.30.1.60
!
no ip domain lookup
!
application
service EA flash:ea/app-cme-ea-2.0.0.0.tcl
paramspace english index 0
paramspace english language en
param ea-password 1234
paramspace english location flash:ea/
paramspace english prefix en
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed 100
no keepalive
!
interface GigabitEthernet0/0.21
encapsulation dot1Q 21
ip address 172.21.200.200 255.255.0.0
ip http server
!
control-plane
!
dial-peer voice 999 voip
service EA out-bound
destination-pattern 0999
session target ipv4:172.21.200.200
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
telephony-service
extension-assigner tag-type provision-tag
max-ephones 51
max-dn 51
ip source-address 172.21.200.200 port 2000
auto-reg-ephone
auto assign 101 to 105
policy-list writeconfig
!
kron policy-list writeconfig
cli write
!
line con 0
line aux 0
line vty 0 4
logging synchronous
!
no scheduler max-task-time
scheduler allocate 20000 1000
!
end
Additional References
The following sections provide references related to extension assigner.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified Communications Express System Administrator
Guide
• Cisco Unified Communications Express Command Reference
Cisco IOS voice configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Voice Command Reference
• Cisco IOS Debug Command Reference
• Cisco IOS Tcl IVR and VoiceXML Application Guide
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Technical Support & Documentation http://www.cisco.com/techsupport
website contains thousands of pages of searchable
technical content, including links to products,
technologies, solutions, technical tips, and tools.
Registered Cisco.com users can log in from this page to
access even more content.
Note Table 9 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes how to generate configuration files for Cisco Unified IP phones that are
connected to a Cisco Unified Communications Manager Express (Cisco Unified CME) router.
Contents
• Information About Configuration Files, page 245
• How to Generate Configuration Files for Phones, page 247
• Where to Go Next, page 254
• Additional References, page 255
By default, there is one shared XML configuration file located in system:/its/ for all Cisco Unified IP
phones that are running SCCP. For SIP phones directly connected to Cisco Unified CME, an individual
configuration profile is created for each phone and stored in system:/cme/sipphone/.
When an IP phone comes online or is rebooted, it automatically gets information about itself from the
appropriate configuration file.
The Cisco universal application loader for phone firmware files allows you to add additional phone
features across all protocols. To do this, a hunt algorithm searches for multiple configuration files. After
a phone is reset or restarted, the phone automatically selects protocol depending on which matching
configuration file is found first. To ensure that Cisco Unified IP phones download the appropriate
configuration for the desired protocol, SCCP or SIP, you must properly configure the IP phones before
connecting or rebooting the phones. The hunt algorithm searches for files in the following order:
1. CTLSEP<mac> file for a SCCP phone—For example, CTLSEP003094C25D2E.tlv
2. SEP <mac> file for a SCCP phone—For example, SEP003094C25D2E.cnf.xml
3. SIP <mac> file for a SIP phone—For example, SIP003094C25D2E.cnf or gk003069C25D2E
4. XML default file for SCCP phones—For example, SEPDefault.cnf.xmls
5. XML default file for SIP phones—For example, SIPDefault.cnf.
In Cisco Unified CME 4.0 and later for SCCP and in Cisco CME 3.4 and later for SIP, you can designate
one of the following locations in which to store configuration files:
• System (Default)—For SCCP phones, one configuration file is created, stored, and used for all
phones in the system. For SIP phones, an individual configuration profile is created for each phone.
• Flash or slot 0—When flash or slot 0 memory on the router is the storage location, you can create
additional configuration files to be applied per phone type or per individual phone, such as user or
network locales.
• TFTP—When an external TFTP server is the storage location, you can create additional
configuration files to be applied per phone type or per individual phone, which are required for
multiple user and network locales.
Restrictions
• Externally stored and per-phone configuration files are not supported on the Cisco Unified IP Phone
7902G, 7910, 7910G, or 7920, or the Cisco Unified IP Conference Station 7935 and 7936.
• TFTP does not support file deletion. When configuration files are updated, they overwrite any
existing configuration files with the same name. If you change the configuration file location, files
are not deleted from the TFTP server.
• Generating configuration files on flash or slot 0 can take up to a minute, depending on the number
of files being generated.
• For smaller routers such as Cisco 2600 series routers, you must manually enter the squeeze
command to erase files after changing the configuration file location or entering any commands that
trigger the deletion of configuration files. Unless you use the squeeze command, the space used by
the moved or deleted configuration files is not usable by other files.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. create cnf-files
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 create cnf-files Builds the XML configuration files required for IP phones.
Example:
Router(config-telephony)# create cnf-files
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Examples
The following example selects flash memory as the configuration file storage location and per-phone as
the type of configuration files that the system generates.
telephony-service
cnf-file location flash:
cnf-file perphone
SUMMARY STEPS
DETAILED STEPS
CONFIG (Version=4.0(0))
=====================
Version 4.0(0)
Cisco Unified CallManager Express
For on-line documentation please see:
www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
ephone-dn 1
number 5001
huntstop
ephone-dn 2
number 5002
huntstop
call-forward noan 5001 timeout 8
tftp-server system:/its/SEPDEFAULT.cnf
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEP00036B54BB15.cnf.xml
tftp-server system:/its/germany/7960-font.xml alias German_Germany/7960-font.xml
tftp-server system:/its/germany/7960-dictionary.xml alias
German_Germany/7960-dictionary.xml
tftp-server system:/its/germany/7960-kate.xml alias German_Germany/7960-kate.xml
tftp-server system:/its/germany/SCCP-dictionary.xml alias
German_Germany/SCCP-dictionary.xml
tftp-server system:/its/germany/7960-tones.xml alias Germany/7960-tones.xml
Caution If your Cisco Unified CME system supports SCCP and also SIP phones, do not connect your SIP phones
to the network until after you have verified the phone configuration profiles.
Prerequisites
• Cisco Unified CME 3.4 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. file text
5. create profile
6. end
DETAILED STEPS
Example:
Router# configure terminal
SUMMARY STEPS
DETAILED STEPS
Note To generate ASCII text files of the configuration profiles for Cisco Unified IP Phone 7905s and 7905Gs,
Cisco Unified IP Phone 7912s and 7912Gs, Cisco ATA-186s, and Cisco ATA-188s, use the file text
command.
The following is sample output from this command displaying information in the configuration profile
for voice register pool 4.
Router# show voice register profile text 4
Pool Tag: 4
# txt
AutoLookUp:0
DirectoriesUrl:0
…
CallWaiting:1
CallForwardNumber:0
Conference:1
AttendedTransfer:1
BlindTransfer:1
…
SIPRegOn:1
UseTftp:1
UseLoginID:0
UIPassword:0
NTPIP:0.0.0.0
UID:2468
…
image_version: "P0S3-07-4-00";
proxy1_address: "10.1.18.100";
proxy2_address: "";
proxy3_address: "";
proxy4_address: "";
proxy5_address: "";
proxy6_address: "";
proxy1_port: "5060";
proxy2_port: "";
proxy3_port: "";
proxy4_port: "";
proxy5_port: "";
proxy6_port: "";
proxy_register: "1";
time_zone: "EST";
dst_auto_adjust: "1";
dst_start_month: "April";
dst_start_day: "";
dst_start_day_of_week: "Sun";
dst_start_week_of_month: "1";
dst_start_time: "02:00";
dst_stop_month: "October";
dst_stop_day: "";
dst_stop_day_of_week: "Sun";
dst_stop_week_of_month: "8";
dst_stop_time: "02:00";
date_format: "M/D/Y";
time_format_24hr: "0";
local_cfwd_enable: "1";
directory_url: "";
messages_uri: "2000";
services_url: "";
logo_url: "";
stutter_msg_waiting: "0";
sync: "0000200155330856";
telnet_level: "1";
autocomplete: "1";
call_stats: "0";
Domain_Name: "";
dtmf_avt_payload: "101";
dtmf_db_level: "3";
dtmf_inband: "1";
dtmf_outofband: "avt";
dyn_dns_addr_1: "";
dyn_dns_addr_2: "";
dyn_tftp_addr: "";
end_media_port: "32766";
http_proxy_addr: "";
http_proxy_port: "80";
nat_address: "";
nat_enable: "0";
nat_received_processing: "0";
network_media_type: "Auto";
network_port2_type: "Hub/Switch";
outbound_proxy: "";
outbound_proxy_port: "5060";
proxy_backup: "";
proxy_backup_port: "5060";
proxy_emergency: "";
proxy_emergency_port: "5060";
remote_party_id: "0";
sip_invite_retx: "6";
sip_retx: "10";
sntp_mode: "directedbroadcast";
sntp_server: "0.0.0.0";
start_media_port: "16384";
tftp_cfg_dir: "";
timer_invite_expires: "180";
timer_register_delta: "5";
timer_register_expires: "3600";
timer_t1: "500";
timer_t2: "4000";
tos_media: "5";
voip_control_port: "5060";
image_version: "P0S3-07-4-00";
user_info: "phone";
line1_name: "1051";
line1_displayname: "";
line1_shortname: "";
line1_authname: "1051";
line1_password: "ww";
line2_name: "";
line2_displayname: "";
line2_shortname: "";
line2_authname: "";
line2_password: "";
auto_answer: "0";
speed_line1: "";
speed_label1: "";
speed_line2: "";
speed_label2: "";
speed_line3: "";
speed_label3: "";
speed_line4: "";
speed_label4: "";
speed_line5: "";
speed_label5: "";
call_hold_ringback: "0";
dnd_control: "0";
anonymous_call_block: "0";
callerid_blocking: "0";
enable_vad: "0";
semi_attended_transfer: "1";
call_waiting: "1";
cfwd_url: "";
cnf_join_enable: "1";
phone_label: "";
preferred_codec: "g711ulaw";
Where to Go Next
After you generate a configuration file for a Cisco Unified IP phone connected to the
Cisco Unified CME router, you are ready to download the file to the phone to be configured. See
“Resetting and Restarting Phones” on page 257.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
This chapter describes how to reset or restart Cisco Unified IP phones that are connected to
Cisco Unified Communications Manager Express (Cisco Unified CME).
Contents
• Information About Resetting and Restarting Phones, page 257
• How to Reset and Restart Phones, page 258
• Additional References, page 265
Note When rebooting multiple IP phones, it is possible for a conflict to occur if too many phones attempt to
access changed Cisco Unified CME configuration information via TFTP simultaneously.
Prerequisites
• Phones to be rebooted are connected to the Cisco Unified CME router.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
or
ephone phone-tag
4. reset {all [time-interval] | cancel | mac-address mac-address | sequence-all}
or
reset
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
or
or
ephone ephone-tag
Enters ephone configuration mode.
Example:
Router(config)# telephony-service
or
Router(config)# ephone 1
Example:
Router(config-telephony)# end
or
Router(config-ephone)# end
Prerequisites
• Phones to be rebooted are connected to the Cisco Unified CME router.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
or
ephone ephone-tag
4. restart {all [time-interval] | mac-address}
or
restart
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
or
or
ephone ephone-tag
Enters ephone configuration mode.
Example:
Router(config)# telephony-service
or
Router(config)# ephone 1
Step 4 restart {all [time-interval] | mac-address} Performs a fast reboot of the specified phone or all phones
or running SCCP associated with this Cisco Unified CME
router. Does not contact the DHCP server for updated
restart
information.
or
Example:
Router(config-telephony)# restart all Performs a fast reboot of the individual SCCP phone being
or configured.
Router(config-ephone)# restart
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Prerequisites
• Cisco Unified CME 3.4 or later.
• The mode cme command must be enabled in Cisco Unified CME.
• Phones to be rebooted are connected to the Cisco Unified CME router.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
or
voice register pool pool-tag
4. reset
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
or parameters for all supported SIP phones in
voice register pool pool-tag Cisco Unified CME.
or
Example: Enters voice register pool configuration mode to set
Router(config)# voice register global phone-specific parameters for SIP phones
or
Router(config)# voice register pool 1
Step 4 reset Performs a complete reboot of all phones connected to this
router that are running SIP, including contacting the DHCP
and TFTP servers for the latest configuration information.
Example:
Router(config-register-global)# reset or
or
Performs a complete reboot of the individual SIP phone
Router(config-register-pool)# reset being configured.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
or
Router(config-register-pool)# end
Prerequisites
• Cisco Unified CME 4.1 or later.
• The mode cme command must be enabled in Cisco Unified CME.
• Phones to be rebooted are connected to the Cisco Unified CME router.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
or
voice register pool pool-tag
4. restart
5. end
DETAILED STEPS
Example:
Router# configure terminal
SUNNARY STEPS
DETAILED STEPS
Step 1 Test local phone operation. Make calls between phones on the Cisco Unified CME router.
Step 2 Place a call from a phone in Cisco Unified CME to a number in the local calling area.
Step 3 Place a call to a phone in Cisco Unified CME from a phone outside this Cisco Unified CME system.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
This chapter describes features that enable Cisco Unified Communications Manager Express
(Cisco Unified CME) to expand or manipulate internal extension numbers so that they conform to
numbering plans used by external systems.
Contents
• Information About Dialing Plans, page 267
• How to Configure Dialing Plans, page 271
• Configuration Examples for Dialing Plan Features, page 284
• Additional References, page 285
• Feature Information for Dialing Plan Features, page 286
In addition, your selection of a numbering scheme for phones that can be directly dialed from the PSTN
is limited by your need to use the range of extensions that are assigned to you by the telephone company
that provides your connection to the PSTN. For example, if your telephone company assigns you a range
from 408 555-0100 to 408 555-0199, you may assign extension numbers only in the range 100 to 199 if
those extensions are going to have Direct Inward Dialing (DID) access. For more information about DID,
see the “Direct Inward Dialing Trunk Lines” section on page 270.
Dial-Plan Patterns
A dial-plan pattern enables abbreviated extensions to be expanded into fully qualified E.164 numbers.
Use dial-plan patterns when configuring a network with multiple Cisco Unified CMEs to ensure that the
appropriate calling number, extension or E.164 number, is provided to the target Cisco Unified CME,
and appears on the phone display of the called phone. In networks that have a single router, you do not
need to use dial-plan patterns.
.When you define a directory number for an SCCP phone, the Cisco Unified CME system automatically
creates a POTS dial peer with the ephone-dn endpoint as a destination. For SIP phones connected
directly into Cisco Unified CME, the dial peer is automatically created when the phone registers. By
default, Cisco Unified CME creates a single POTS dial peer for each directory number.
For example, when the ephone-dn with the number 1001 was defined, the following POTS dial peer was
automatically created for it:
dial-peer voice 20001 pots
destination-pattern 1001
voice-port 50/0/2
A dial-plan pattern builds additional dial peers for the expanded numbers it creates. If a dialplan pattern
is configured and it matches against a directory number, two POTS dial peers are created, one for the
abbreviated number and one for the complete E.164 direct-dial telephone number.
For example, if you then define a dial-plan pattern that 1001 will match, such as 40855500.., a second
dial peer is created so that calls to both the 0001 and 4085550001 numbers are completed. In this
example, the additional dial peer that is automatically created looks like the following:
dial-peer voice 20002 pots
destination-pattern 40855510001
voice-port 50/0/2
In networks with multiple routers, you may need to use dial-plan patterns to expand extensions to E.164
numbers because local extension numbering schemes can overlap each other. Networks with multiple
routers have authorities such as gatekeepers that route calls through the network. These authorities
require E.164 numbers so that all numbers in the network are unique. Define dial-plan patterns to expand
extension numbers into unique E.164 numbers for registering with a gatekeeper.
If multiple dial-plan patterns are defined, the system matches extension numbers against the patterns in
sequential order, starting with the lowest numbered dial-plan pattern tag first. Once a pattern matches an
extension number, the pattern is used to generate an expanded number. If additional patterns
subsequently match the extension number, they are not used.
Dial-Plan Patterns
• SCCP: Configuring Dial-Plan Patterns, page 271 (required)
• SIP: Configuring Dial-Plan Patterns, page 272 (required)
• Verifying Dial-Plan Patterns, page 274 (optional)
Tip In networks that have a single router, you do not need to define dial-plan patterns.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. dialplan-pattern tag pattern extension-length extension-length [extension-pattern
extension-pattern | no-reg]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 dialplan-pattern tag pattern extension-length length Maps a digit pattern for an abbreviated
[extension-pattern epattern] [no-reg] extension-number prefix to the full E.164 telephone
number pattern.
Example:
Router(config-telephony)# dialplan-pattern 1
4085550100 extension-length 3 extension-pattern 4..
Prerequisites
Cisco Unified CME 4.0 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 dialplan-pattern tag pattern extension-length Defines pattern that is used to expand abbreviated extension
extension-length [extension-pattern numbers of SIP calling numbers in Cisco Unified CME into
extension-pattern | no-reg]
fully qualified E.164 numbers.
Example:
Router(config-register-global)#
dialplan-pattern 1 4085550... extension-length
5
Step 5 call-forward system redirecting-expanded Applies dial-plan pattern expansion globally to redirecting,
including originating and last reroute, numbers for SIP
extensions in Cisco Unified CME for call forward using
Example:
Router(config-register-global)# call-forward
B2BUA.
system redirecting-expanded
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
SUMMARY STEPS
1. show telephony-service
2. show telephony-service dial-peer
or
show dial-peer summary
DETAILED STEPS
Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions
To define voice translation rules and voice translation profiles, perform the following steps.
Note To configure translation rules for voice calls in Cisco CME 3.1 and earlier versions, see the Cisco IOS
Voice, Video, and FAX Configuration Guide.
Prerequisites
Cisco CME 3.2 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice translation-rule number
4. rule precedence /match-pattern/ /replace-pattern/
5. exit
6. voice translation-profile name
7. translate {called | calling | redirect-called} voice-translation-rule-tag
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice translation-rule number Defines a translation rule for voice calls and enters voice
translation-rule configuration mode.
Example: • number—Number that identifies the translation rule.
Router(config)# voice translation-rule 1 Range is 1 to 2147483647.
Example:
Router(cfg-translation-rule)# exit
Step 6 voice translation-profile name Defines a translation profile for voice calls.
• name—Name of the translation profile. Maximum
Example: length of the voice translation profile name is
Router(config)# voice translation-profile name1 31 alphanumeric characters.
Step 7 translate {called | calling | redirect-called} Associates a voice translation rule with a voice translation
voice-translation-rule-tag profile.
• called—Associates the translation rule with called
Example: numbers.
Router(cfg-translation-profile)# translate
called 1 • calling—Associates the translation rule with calling
numbers.
• redirect-called—Associates the translation rule with
redirected called numbers.
• translation-rule-tag—Reference number of the
translation rule. Range is 1 to 2147483647.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(cfg-translation-profile)# end
What to Do Next
• To apply voice translation profiles to SCCP phones connected to Cisco Unified CME 3.2 or a later
version, see the “SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions”
section on page 277.
• To apply voice translation profiles to SIP phones connected to Cisco Unified CME 4.1 or a later
version, see the “SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later”
section on page 280.
• To apply voice translation profiles to SIP phones connected to Cisco Unified CME 3.4 or
Cisco Unified 4.0(x), see the “SIP: Applying Voice Translation Rules before Cisco Unified CME
4.1” section on page 281.
SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions
To apply a voice translation profile to modify the number dialed by extensions on a SCCP phone,
perform the following steps.
Prerequisites
• Cisco CME 3.2 or a later version.
• Voice translation profile containing voice translation rules to be applied must be already configured.
For configuration information, see the “Defining Voice Translation Rules in Cisco CME 3.2 and
Later Versions” section on page 275.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn tag
4. translation-profile {incoming | outgoing} name
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn tag Enters ephone-dn configuration mode to create an
extension (ephone-dn) for a Cisco Unified IP phone line,
an intercom line, a paging line, a voice-mail port, or a
Example:
Router(config)# ephone-dn 1
message-waiting indicator (MWI).
• tag—Unique sequence number that identifies this
ephone-dn during configuration tasks. Range is
1 to the maximum number of ephone-dns allowed on
the router platform. See the CLI help for the maximum
value for this argument.
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
Prerequisites
Translation rule to be applied must be already configured by using the translation-rule and rule
commands. For configuration information, see the Cisco IOS Voice, Video, and FAX Configuration
Guide.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. translate {called | calling} translation-rule-number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn tag Enters ephone-dn configuration mode to create directory
number for a Cisco Unified IP phone line, an intercom line,
a paging line, a voice-mail port, or a message-waiting
Example:
Router(config)# ephone-dn 1
indicator (MWI).
Step 4 translate {called | calling} Specifies rule to be applied to the directory number being
translation-rule-tag configured.
• translation-rule-tag—Reference number of previously
Example: configured translation rule. Range: 1 to 2147483647.
Router(config-ephone-dn)# translate called 1
• You can use an ephone-dn template to apply this
command to one or more directory numbers. If you use
an ephone-dn template to apply a command to a
directory number and you also use the same command
in ephone-dn configuration mode for the same
directory number, the value that you set in ephone-dn
configuration mode has priority.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(cfg-translation-profile)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later
To apply a voice translation profile for incoming call legs to a directory number on a SIP phone, perform
the following steps.
Prerequisites
• Cisco Unified CME 4.1 or a later version.
• Voice translation profile containing voice translation rules to be applied must be already configured.
For configuration information, see the “Defining Voice Translation Rules in Cisco CME 3.2 and
Later Versions” section on page 275.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. translation-profile incoming name
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice
port, or a message-waiting indicator (MWI).
Example:
Router(config-register-dn)# ephone-dn 1
Step 4 translation-profile incoming name Assigns a translation profile for incoming call legs to this
directory number.
Example:
Router(config-register-dn)# translation-profile
incoming name1
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone-dn)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “SIP: Generating Configuration Profiles for SIP Phones” on page 250.
SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1
To apply an already-configured voice translation rule to modify the number dialed by extensions on a
SIP phone, perform the following steps.
Prerequisites
• Cisco CME 3.4 or a later version.
• Voice translation rule to be applied must be already configured. For configuration information, see
the “Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions” section on page 275.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool tag
4. translate-outgoing {called | calling} rule-tag
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example:
Router(config)# voice register pool 3
Step 4 translate-outgoing {called | calling} rule-tag Specifies an already configured voice translation rule to be
applied to SIP phone being configured.
Example:
Router(config-register-pool)#
translate-outgoing called 1
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “SIP: Generating Configuration Profiles for SIP Phones” on page 250.
SUMMARY STEPS
DETAILED STEPS
Translation-rule tag: 6
Rule 1:
Match pattern: 65088801..
Replace pattern: 6508880101
Match type: none Replace type: none
Match plan: none Replace plan: none
Prerequisite
• Cisco CME 3.0 or a later version.
• PSTN access prefix must be configured for outbound dial peer.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. secondary-dialtone digit-string
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 11 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the localization support in Cisco Unified Communications Manager Express
(Cisco Unified CME) for languages other than English and network tones and cadences not specific to
the United States.
Contents
• Information About Localization, page 287
• How to Configure Localization Support, page 289
• Configuration Examples for Localization, page 299
• Where to Go Next, page 301
• Additional References, page 301
• Feature Information for Localization Support, page 302
System-Defined Locales
Cisco Unified CME provides internal localization support for 12 languages including English and
16 countries including the United States. User locales specify the language to use for text displays;
network locales specify country-specific tones and cadences. View the list of system-defined locales and
their two-letter codes by typing ? after the user-locale and network-locale commands in
telephony-service configuration mode.
For the Cisco Unified IP Phone 7912, 7940, and 7960, the system-defined user locales and network
locales are preloaded into Cisco IOS software.
For the Cisco Unified IP Phone 7911, 7941, 7961, 7970, and 7971, you must download locale files to
support the system-defined locales and store the files in slot 0, flash memory, or on an external TFTP
server. See the “Installing System-Defined Locales for Cisco Unified IP Phone 7911, 7941, 7961, 7970,
and 7971” section on page 289.
User-Defined Locales
You may need to support locales other than the system-defined locales that are predefined in the system.
The user-defined locale feature allows you to specify user and network locales in addition to those that
are predefined in Cisco IOS software and apply them to individual ephones. For example, if your site
has phones that must use the language and tones for Traditional Chinese, which is not one of the
system-defined choices, you must install the locale files for Traditional Chinese.
In Cisco Unified CME 4.0 and later, you can download and install the files to support a particular user
and network locale in flash, slot 0, or an external TFTP server. You cannot install these files in the
system location. These user and network locales can then be assigned to all or some phones.
User-defined language codes for user locales are based on ISO 639 codes, which are available at the
Library of Congress website: http://www.loc.gov/standards/iso639-2/. User-defined country codes for
network locales are based on ISO 3166 codes.
For configuration information, see the “Installing User-Defined Locales” section on page 292.
Multiple Locales
In Cisco Unified CME 4.0 and later, you can specify up to five user and network locales and apply
different locales to individual ephones or groups of ephones using ephone templates. For example, you
can specify French for phones A, B, and C; German for phones D, E, and F; and English for phones G,
H, and I. Only one user and network locale can be applied to each phone.
Each of the five user and network locales that you can define in a multilocale system is identified by a
locale tag. The locale identified by tag 0 is always the default locale, although you can define this default
to be any supported locale. For example, if you define user locale 0 to be JP (Japanese), the default user
locale for all phones is JP. If you do not specify a locale for tag 0, the default is US (United States).
To apply alternative locales to different phones, you must use per-phone configuration files to build
individual configuration files for each phone. The configuration files automatically use the default
user-locale 0 and network-locale 0. You can override these defaults for individual phones by configuring
alternative locale codes and then creating ephone-templates to assign the locales to individual ephones.
For configuration information, see the “Configuring Multiple Locales” section on page 295.
Installing System-Defined Locales for Cisco Unified IP Phone 7911, 7941, 7961,
7970, and 7971
To install locale files for system-defined locales, perform the following steps.
Prerequisites
• Cisco Unified CME 4.0(2) or a later version.
• You must create per-phone configuration files as described in the “SCCP: Defining Per-Phone
Configuration Files and Alternate Location” section on page 129.
Restrictions
• Cisco Unified IP Phone 7931G supports United States English only.
Step 1 Go to http://www.cisco.com/cgi-bin/tablebuild.pl/CME-Locale
You must have an account on Cisco.com to access the Software Download Center. If you do not have
an account or have forgotten your username or password, click Cancel at the login dialog box and follow
the instructions that appear.
Step 2 Select your version of Cisco Unified CME.
Step 3 Select the TAR file for the locale you want to install. Each TAR file contains locale files for a specific
language and country and uses the following naming convention:
CME-locale-language_country-CMEversion
For example, CME-locale-de_DE-4.0.2-2.0 is German for Germany for Cisco Unified CME 4.0(2).
Step 4 Download the TAR file to a TFTP server that is accessible to the Cisco Unified CME router. Each file
contains all the firmware required for all phone types supported by that version of Cisco Unified CME.
Step 5 Use the archive tar command to extract the files to flash, slot 0, or an external TFTP server.
Router# archive tar /xtract source-url flash:/file-url
For example, to extract the contents of CME-locale-de_DE-4.0.2-2.0.tar from TFTP server 192.168.1.1
to router flash memory, use this command:
Router# archive tar /xtract tftp://192.168.1.1/cme-locale-de_DE-4.0.2-2.0.tar flash:
Step 6 See Table 12 and Table 13 for a description of the codes used in the filenames and the list of supported
directory names.
Each phone type has a JAR file that uses the following naming convention:
language-phone-sccp.jar
For example, de-td-sccp.jar is for German on the Cisco Unified IP Phone 7970.
Each TAR file also includes the file g3-tones.xml for country-specific network tones and cadences.
Step 7 If you store the locale files in flash or slot 0: on the Cisco Unified CME router, create a TFTP alias for
the user locale (text displays) and network locale (tones) using this format:
Router(config)# tftp-server flash:/jar_file alias directory_name/td-sccp.jar
Router(config)# tftp-server flash:/g3-tones.xml alias directory_name/g3-tones.xml
Use the appropriate directory name shown in Table 13 and remove the two-letter language code from
the JAR file name.
For example, the TFTP aliases for German and Germany for the Cisco Unified IP Phone 7970 are:
Router(config)# tftp-server flash:/de-td-sccp.jar alias German_Germany/td-sccp.jar
Router(config)# tftp-server flash:/g3-tones.xml alias Germany/g3-tones.xml
Note On Cisco 3800 series routers, you must include /its in the directory name (flash:/its or slot0:/its). For
example, the TFTP alias for German for the Cisco Unified IP Phone 7970 is:
Router# tftp-server flash:/its/de-td-sccp.jar alias German_Germany/td-sccp.jar
Step 8 If you store the locale files on an external TFTP server, create a directory under the TFTP root directory
for each user and network locale.
Use the appropriate directory name shown in Table 13 and remove the two-letter language code from
the JAR file name.
For example, the user-locale directory for German and the network-locale directory for Germany for the
Cisco Unified IP Phone 7970 are:
TFTP-Root/German_Germany/td-sccp.jar
TFTP-Root/Germany/g3-tones.xml
Step 9 For Russian and Japanese, you must copy the UTF8 dictionary file into flash to use special phrases.
• Only flash can be used for these locales. Copy russian_tags_utf8_phrases for Russian;
Japanese_tags_utf8_phrases for Japanese.
• Use the user-locale jp and user-locale ru command to load the UTF8 phrases into
Cisco Unified CME.
Step 10 Assign the locales to phones. To set a default locale for all phones, use the user-locale and
network-locale commands in telephony-service configuration mode.
Step 11 To support more than one user or network locale, see the “Configuring Multiple Locales” section on
page 295.
Step 12 Use the create cnf-files command to rebuild the configuration files.
Step 13 Use the reset command to reset the phones and see the localized displays.
Prerequisites
• Cisco Unified CME 4.0(3) or a later version.
• You must create per-phone configuration files as described in the “SCCP: Defining Per-Phone
Configuration Files and Alternate Location” section on page 129.
Restrictions
• User-defined locales are not supported on the Cisco Unified IP Phone 7920 or 7936.
• User-defined locales are not supported if the configuration file location is system.
• When you use the setup tool from the telephony-service setup command to provision phones, you
can only choose a default user locale and network locale, and you are limited to selecting a locale
code that is supported in the system. You cannot use multiple locales or user-defined locales with
the setup tool.
• When using a user-defined locale, the phone normally displays text using the user-defined fonts,
except for any strings that are interpreted by Cisco Unified CME, such as “Cisco/Personal
Directory,” “Speed Dial/Fast Dial,” and so forth.
Step 1 Go to http://www.cisco.com/cgi-bin/tablebuild.pl/CME-Locale
Step 2 Select your version of Cisco Unified CME.
Step 3 Select the TAR file for the locale that you want to install. Each TAR file contains locale files for a
specific language and country and uses the following naming convention:
CME-locale-language_country-CMEversion-fileversion
For example, CME-locale-zh_CN-4.0.3-2.0 is Traditional Chinese for China for
Cisco Unified CME 4.0(3).
Step 4 Download the TAR file to a TFTP server that is accessible to the Cisco Unified CME router. Each file
contains all the firmware required for all phone types supported by that version of Cisco Unified CME.
Step 5 Use the archive tar command to extract the files to slot 0, flash, or an external TFTP server.
Router# archive tar /xtract source-url flash:/file-url
For example, to extract the contents of CME-locale-zh_CN-4.0.3-2.0.tar from TFTP server 192.168.1.1
to router flash memory, use this command:
Router# archive tar /xtract tftp://192.168.1.1/cme-locale-zh_CN-4.0.3-2.0.tar flash:
Step 6 For Cisco Unified IP Phones 7905, 7912, 7940, or 7960, go to Step 11.
For Cisco Unified IP Phones 7911, 7941, 7961, 7970, or 7971, go to Step 7.
Step 7 Each phone type has a JAR file that uses the following naming convention:
language-type-sccp.jar
For example, zh-td-sccp.jar is Traditional Chinese for the Cisco Unified IP Phone 7970.
See Table 14 and Table 15 for a description of the codes used in the filenames.
Step 8 If you store the locale files in flash or slot 0: on the Cisco Unified CME router, create a TFTP alias using
this format:
Router(config)# tftp-server flash:/jar_file alias directory_name/td-sccp.jar
Remove the two-letter language code from the JAR filename and use one of five supported directory
names with the following convention:
user_define_number, where number is 1 to 5
For example, the alias for Chinese on the Cisco Unified IP Phone 7970 is:
Router(config)# tftp-server flash:/zh-td-sccp.jar alias user_define_1/td-sccp.jar
Note On Cisco 3800 series routers, you must include /its in the directory name (flash:/its or slot0:/its). For
example, the TFTP alias for Chinese for the Cisco Unified IP Phone 7970 is:
Router(config)# tftp-server flash:/its/zh-td-sccp.jar alias user_define_1/td-sccp.jar
Step 9 If you store the locale files on an external TFTP server, create a directory under the TFTP root directory
for each locale.
Remove the two-letter language code from the JAR filename and use one of five supported directory
names with the following convention:
user_define_number, where number is 1 to 5
For example, for Chinese on the Cisco Unified IP Phone 7970, remove “zh” from the JAR filename and
create the “user_define_1” directory under TFTP-Root on the TFTP server:
TFTP-Root/user_define_1/td-sccp.jar
Step 10 Go to Step 14.
Step 11 Download one or more of the following XML files depending on your selected locale and phone type.
All required files are included in the JAR file.
7905-dictionary.xml
7905-font.xml
7905-kate.xml
7920-dictionary.xml
7960-dictionary.xml
7960-font.xml
7960-kate.xml
7960-tones.xml
SCCP-dictionary.utf-8.xml
SCCP-dictionary.xml
Step 12 Rename these files and copy them to flash, slot 0, or an external TFTP server. Rename the files using
the format user_define_number_filename where number is 1 to 5. For example, use the following names
if you are setting up the first user-locale:
user_define_1_7905-dictionary.xml
user_define_1_7905-font.xml
user_define_1_7905-kate.xml
user_define_1_7920-dictionary.xml
user_define_1_7960-dictionary.xml
user_define_1_7960-font.xml
user_define_1_7960-kate.xml
user_define_1_7960-tones.xml
user_define_1_SCCP-dictionary.utf-8.xml
user_define_1_SCCP-dictionary.xml
Step 13 If you store the locale files in flash or slot 0: on the Cisco Unified CME router, create a TFTP alias, for
example:
tftp-server flash:user_define_1_7905-dictionary.xml
tftp-server flash:user_define_1_7905-font.xml
tftp-server flash:user_define_1_7905-kate.xml
tftp-server flash:user_define_1_7960-tones.xml
tftp-server flash:user_define_1_7960-dictionary.xml
tftp-server flash:user_define_1_7960-font.xml
tftp-server flash:user_define_1_7960-kate.xml
tftp-server flash:user_define_1_SCCP-dictionary.utf-8.xml
tftp-server flash:user_define_1_SCCP-dictionary.xml
Step 14 Copy the language_tags_file and language_utf8_tags_file to the location of the other locale files (flash,
slot 0, or TFTP server). Rename the files to user_define_number_tags_file and
user_define_number_utf8_tags_file respectively, where number is 1 to 5 and matches the user-defined
directory.
Step 15 Assign the locales to phones. See the “Configuring Multiple Locales” section on page 295.
Step 16 Use the create cnf-files command to rebuild the configuration files.
Step 17 Use the reset command to reset the phones and see the localized displays.
Prerequisites
• Cisco Unified CME 4.0 or a later version.
• To specify alternative user and network locales for individual phones in a Cisco Unified CME
system, you must use per-phone configuration files. For more information, see the “SCCP: Defining
Per-Phone Configuration Files and Alternate Location” section on page 129.
• You can also use user-defined locale codes as alternative locales after you download the appropriate
XML files. See the“Installing User-Defined Locales” section on page 292.
Restrictions
• Multiple user and network locales are not supported on the Cisco Unified IP Phone 7902G, 7910,
7910G, or 7920, or the Cisco Unified IP Conference Station 7935 and 7936.
• When you use the setup tool from the telephony-service setup command to provision phones, you
can only choose a default user locale and network locale, and you must select a locale code that is
predefined in the system. You cannot use multiple or user-defined locales with the setup tool.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. user-locale user-locale-tag [user-defined-code] language-code
5. network-locale network-locale-tag [user-defined-code] country-code
6. create cnf-files
7. exit
8. ephone-template template-tag
9. user-locale user-locale-tag
10. network-locale network-locale-tag
11. exit
12. ephone phone-tag
13. ephone-template template-tag
14. exit
15. telephony service
16. reset {all [time-interval] | cancel | mac-address mac-address | sequence-all}
17. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 user-locale user-locale-tag [user-defined-code] Specifies a language for phone displays.
language-code
• user-locale-tag—Assigns a locale identifier to the
language code. Range is 0 to 4. This argument is
Example: required when using multiple locales; otherwise the
Router(config-telephony)# user-locale 1 U1 ZH specified language is the default applied to all phones.
• user-defined-code—(Optional) Assigns one of the
user-defined codes to the specified language code.
Valid codes are U1, U2, U3, U4, and U5.
• language-code—Type ? to display a list of
system-defined codes. United States (US) is the default.
You can assign any valid ISO 639 code to a
user-defined code (U1 to U5).
Step 5 network-locale network-locale-tag Specifies a country for tones and cadences.
[user-defined-code] country-code
• network-locale-tag—Assigns a locale identifier to the
country code. Range is 0 to 4. This argument is required
Example: when using multiple locales; otherwise the specified
Router(config-telephony)# network-locale 1 FR country is the default applied to all phones.
• user-defined-code—(Optional) Assigns one of the
user-defined codes to the specified country code. Valid
codes are U1, U2, U3, U4, and U5.
• country-code—Type ? to display a list of
system-defined codes. United States (US) is the default.
You can assign any valid ISO 3166 code to a
user-defined code (U1 to U5).
Step 6 create cnf-files Builds the required XML configuration files for IP phones.
Use this command after you update configuration file
parameters such as the user locale or network locale.
Example:
Router(config-telephony)# create cnf-files
Example:
Router(config-telephony)# exit
Step 8 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique sequence number that identifies
Example: this template during configuration tasks.
Router(config)# ephone template 1
Step 9 user-locale user-locale-tag Assigns a user locale to this ephone template.
• user-locale-tag—A locale tag that was created in
Example: Step 4. Range is 0 to 4.
Router(config-ephone-template)# user-locale 2
Step 10 network-locale network-locale-tag Assigns a network locale to this ephone template.
• network-locale-tag—A locale tag that was created in
Example: Step 5. Range is 0 to 4.
Router(config-ephone-template)#
network-locale 2
Step 11 exit Exits ephone-template configuration mode.
Example:
Router(config-ephone-template)# exit
Step 12 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks.
Router(config)# ephone 36
Step 13 ephone-template template-tag Applies an ephone template to an ephone.
• template-tag—Number of the template to apply to this
Example: ephone.
Router(config-ephone)# ephone-template 1
Step 14 exit Exits ephone configuration mode.
Example:
Router(config-ephone)# exit
Step 15 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
tftp-server system:/its/SEPDEFAULT.cnf
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEP00036B54BB15.cnf.xml
tftp-server system:/its/germany/7960-font.xml alias German_Germany/7960-font.xml
tftp-server system:/its/germany/7960-dictionary.xml alias
German_Germany/7960-dictionary.xml
tftp-server system:/its/germany/7960-kate.xml alias German_Germany/7960-kate.xml
tftp-server system:/its/germany/SCCP-dictionary.xml alias
German_Germany/SCCP-dictionary.xml
tftp-server system:/its/germany/7960-tones.xml alias Germany/7960-tones.xml
Step 2 Ensure that per-phone configuration files are defined with the cnf-file perphone command.
Step 3 Use the show telephony-service ephone-template command to check the user locale and network
locale settings in each ephone template.
Step 4 Use the show telephony-service ephone command to check that the correct templates are applied to
phones.
Step 5 Use the debug tftp events command to see which files Cisco Unified CME is looking for and whether
the files are found and opened correctly. There are usually three states (“looking for x file” “opened x
file” and “finished x file”). The file is found when all three states are displayed. Use this command if
the configuration file location is not TFTP. For an external TFTP server you can use the logs from the
TFTP server.
After using the previous commands to define Germany as the default user and network locale, use the
following commands to return the default value of 0 to US:
telephony service
no user-locale 0 DE
no network-locale 0 DE
Another way to define Germany as the default user and network locale is to use the following commands:
telephony service
cnf-file location flash:
cnf-file perphone
user-locale DE
network-locale DE
After using the previous commands, use the following commands to return the default to US:
telephony service
no user-locale DE
no network-locale DE
The following example defines three alternative locales: JP (Japan), FR (France), and ES (Spain). The
default is US for all phones that do not have an alternative applied using ephone templates. In this
example, ephone 11 uses JP for its locales, ephone 12 uses FR, ephone 13 uses ES, and ephone 14 uses
the default, US.
telephony-service
cnf-file location flash:
cnf-file perphone
create cnf-files
user-locale 1 JP
user-locale 2 FR
user-locale 3 ES
network-locale 1 JP
network-locale 2 FR
network-locale 3 ES
create cnf-files
ephone-template 1
user-locale 1
network-locale 1
ephone-template 2
user-locale 2
network-locale 2
ephone-template 3
user-locale 3
network-locale 3
ephone 11
button 1:25
ephone-template 1
ephone 12
button 1:26
ephone-template 2
ephone 13
button 1:27
ephone-template 3
ephone 14
button 1:28
ephone-template 2
user-locale 1
network-locale 1
ephone 11
button 1:25
ephone-template 2
ephone 12
button 1:26
Where to Go Next
Ephone Templates
For more information about ephone templates, see “Creating Templates” on page 881.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 16 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
Cisco Unified
Feature Name CME Version Feature Information
Multiple Locales 4.0 Support for multiple user and network locales was
introduced.
User-Defined Locales 4.0 User-defined locales were introduced.
This chapter describes the transcoding support available in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Prerequisites for Transcoding Resources, page 303
• Restrictions for Transcoding Resources, page 304
• Information About Transcoding Resources, page 304
• How to Configure Transcoding Resources, page 306
• Configuration Examples for Transcoding Resources, page 333
• Where to go Next, page 335
• Additional References, page 335
• Feature Information for Transcoding Resources, page 337
Transcoding Support
Transcoding compresses and decompresses voice streams to match endpoint-device capabilities.
Transcoding is required when an incoming voice stream is digitized and compressed (by means of a
codec) to save bandwidth, and the local device does not support that type of compression.
Cisco CME 3.2 and later versions support transcoding between G.711 and G.729 codecs for the
following features:
• Ad hoc conferencing—One or more remote conferencing parties uses G.729.
• Call transfer and forward—One leg of a Voice over IP (VoIP)-to-VoIP hairpin call uses G.711 and
the other leg uses G.729. A hairpin call is an incoming call that is transferred or forwarded over the
same interface from which it arrived.
• Cisco Unity Express—An H.323 or SIP call using G.729 is forwarded to Cisco Unity Express.
Cisco Unity Express supports only G.711, so G.729 must be transcoded. See the
Cisco Unity Express documentation at
www.cisco.com/en/US/products/sw/voicesw/ps5520/tsd_products_support_series_home.html
• Music on hold (MOH)—The phone receiving MOH is part of a system that uses G.729. The G.711
MOH is transcoded into G.729 resulting in a poorer quality sound due to the lower compression of
G.729.
Figure 14 provides an example of each of the four call situations described.
Figure 14 Three-Way Conferencing, Call Transfer and Forward, Cisco Unity Express, and MOH
Between G.711 and G.729
Conferencing
Phone A calls phone B.
PSTN C
Phone B conferences phone C.
Call Transfer and Forward
Phone A calls phone B.
Phone B transfers or forwards Branch office
to phone C.
PSTN gateway IP
V
A IP
G.711
IP Central Office Branch office B
IP
IP WAN IP
G.729 G.729
Cisco 3745 Cisco 2800
with PVDM2, CME, IP
CUE
IP MOH, and CUE 50 phones
Phone A calls phone B using H.323 or SIP.
120 phones Phone B is busy and phone A is sent to voice mail.
MOH
103375
Phone A calls phone B.
Phone B answers and places phone A on hold.
used after the voice path is established with transcoding. However, during the SCCP manipulations, a
temporary session may be allocated. If this temporary session cannot be allocated, the transcoding
request is not honored, and the call continues with the G.711 codec.
If the codec g729r8 dspfarm-assist command is configured for a phone and a DSP resource is not
available when needed for transcoding, a phone registered to the local Cisco Unified CME router will
use G.711 instead of G.729r8. This is not true for nonSCCP call legs; if DSP resources are not available
for the transcoding required for a conference, for example, the conference is not created.
0
1 DSP DSP DSP
2 DSP DSP DSP
3 DSP DSP DSP
4 DSP DSP DSP
DSP DSP DSP
103376
PVDM slots
or SIMM socket
Use DSP resources to provide voice termination of the digital voice trunk group or resources for a DSP
farm. DSP resources available for transcoding and not used for voice termination are referred to as a DSP
farm. Figure 16 shows a DSP farm managed by Cisco Unified CME.
DSP = Transcoding
DSP DSP DSP
DSP = Voice termination
103378
Note Transcoding of G.729 calls to G.711 allows G.729 calls to participate in existing G.711 software-based,
three-party conferencing, thus eliminating the need to divide DSPs between transcoding and
conferencing.
To determine how many DSP voice resources are on your Cisco Unified CME router, use the show voice
dsp command. To determine how many DSP farms have been configured, use the show sdspfarm
sessions and show sdspfarm units commands. For more information about these commands, see the
Cisco Unified Communications Manager Express Command Reference.
For information on determining if your router has the correct DSP allocation for transcoding, see the
“Allocation of DSP Resources” section in the “Configuring Enhanced Conferencing and Transcoding for
Voice Gateway Routers” chapter of the Cisco Unified Communications Manager and
Cisco IOS Interoperability Guide.
What to Do Next
Choose from one of the following options:
• To set up DSP farms for NM-HDVs, see the “Configuring DSP Farms for NM-HDVs” section on
page 314.
• To set up DSP farms on NM-HDs and NM-HDV2s, see the “Modifying the Number of Transcoding
Sessions for NM-HDVs” section on page 316.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice-card slot
4. dsp services dspfarm
5. exit
6. sccp local interface-type interface-number
7. sccp ccm ip-address identifier identifier-number
8. sccp
9. sccp ccm group group-number
10. bind interface interface-type interface-number
11. associate ccm identifier-number priority
12. associate profile profile-identifier register device-name
13. keepalive retries number
14. switchover method {graceful | immediate}
15. switchback method {graceful | guard timeout-guard-value | immediate | uptime
uptime-timeout-value}
16. switchback interval seconds
17. exit
18. dspfarm profile profile-identifier transcode [security]
19. trustpoint trustpoint-label
20. codec codec-type
21. maximum sessions number
22. associate application sccp
23. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice-card slot Enters voice-card configuration mode and identifies the slot
in the chassis in which the NM-HDV or NM-HDV farm is
located.
Example:
Router(config)# voice-card 1
Step 4 dsp services dspfarm Enables DSP-farm services on the NM-HDV or NM-HDV
farm.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 5 exit Exits voice-card configuration mode.
Example:
Router(config-voicecard)# exit
Step 6 sccp local interface-type interface-number Selects the local interface that the SCCP applications
(transcoding and conferencing) should use to register with
Cisco Unified CME.
Example:
Router(config)# sccp local FastEthernet 0/0 • interface-type—Interface type that the SCCP
application uses to register with Cisco Unified CME.
The type can be an interface address or a
virtual-interface address such as Ethernet.
• interface-number—Interface number that the SCCP
application uses to register with Cisco Unified CME.
Step 7 sccp ccm ip-address identifier Specifies the Cisco Unified CME address.
identifier-number
• ip-address—IP address of the Cisco Unified CME
server.
Example:
Router(config)# sccp ccm 10.10.10.1 priority 2
• identifier identifier-number—Identifier used to
associate the SCCP Cisco Unified CME IP address
with a Cisco Unified CME group. See the associate
ccm command in Step 11.
• Repeat this step to specify the address of a secondary
Cisco Unified CME server.
Step 8 sccp Enables SCCP and its associated transcoding and
conferencing applications.
Example:
Router(config)# sccp
Example:
Router(config-sccp-ccm)# exit
Step 18 dspfarm profile profile-identifier transcode Enters DSP farm profile configuration mode and defines a
[security] profile for DSP farm services.
• profile-identifier—Number that uniquely identifies a
Example: profile. Range is 1 to 65535. There is no default.
Router(config)# dspfarm profile 1 transcode
security • transcode—Enables profile for transcoding.
• security—Enables profile for secure DSP farm
services.
Step 19 trustpoint trustpoint-label (Optional) Associates a trustpoint with a DSP farm profile.
Example:
Router(config-dspfarm-profile)# trustpoint
dspfarm
Example:
Router(config-dspfarm-profile)# associate
application sccp
Step 23 end Returns to privileged EXEC mode.
Example:
Router(config-dspfarm-profile)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. voice-card slot
4. dsp services dspfarm
5. exit
6. sccp local interface-type interface-number
7. sccp ccm ip-address priority priority-number
8. sccp
9. dspfarm transcoder maximum sessions number
10. dspfarm
11. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice-card slot Enters voice-card configuration mode and identifies the slot
in the chassis in which the NM-HDV or NM-HDV farm is
located.
Example:
Router(config)# voice-card 1
Step 4 dsp services dspfarm Enables DSP-farm services on the NM-HDV or NM-HDV
farm.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 5 exit Returns to global configuration mode.
Example:
Router(config-voicecard)# exit
Step 6 sccp local interface-type interface-number Selects the local interface that the SCCP applications
(transcoding and conferencing) should use to register with
Cisco Unified CME.
Example:
Router(config)# sccp local FastEthernet 0/0 • interface-type—Interface type that the SCCP
application uses to register with Cisco Unified CME.
The type can be an interface address or a
virtual-interface address such as Ethernet.
• interface-number—Interface number that the SCCP
application uses to register with Cisco Unified CME.
Step 7 sccp ccm ip-address priority priority-number Specifies the Cisco Unified CME address.
• ip-address—IP address of the Cisco Unified CME
Example: server.
Router(config)# sccp ccm 10.10.10.1 priority 1
• priority priority—Priority of the Cisco Unified CME
server relative to other connected servers. Range is
1 (highest) to 4 (lowest).
Step 8 sccp Enables SCCP and its associated transcoding and
conferencing applications.
Example:
Router(config)# sccp
Example:
Router(config)# dspfarm
Step 11 end Returns to privileged EXEC mode.
Example:
Router(config)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. no dspfarm
4. dspfarm transcoder maximum sessions number
5. dspfarm
6. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# no dspfarm
Step 4 dspfarm transcoder maximum sessions number Specifies the maximum number of transcoding sessions to
be supported by the DSP farm.
Example:
Router(config)# dspfarm transcoder maximum
sessions 12
Step 5 dspfarm Enables the DSP farm.
Example:
Router(config)# dspfarm
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config)# end
Configuring the Cisco Unified CME Router to Act as the DSP Farm Host
To configure the Cisco Unified CME router to act as the DSP farm host, perform the following tasks.
• Determining the Maximum Number of Transcoder Sessions, page 317
• Setting the Cisco Unified CME Router to Receive IP Phone Messages, page 318
• Configuring the Cisco Unified CME Router to Host a Secure DSP Farm, page 320
SUMMARY STEPS
DETAILED STEPS
Step 1 Use the dspfarm transcoder maximum sessions command to set the maximum number of transcoder
sessions you have configured.
Step 2 Use the show sdspfarm sessions command to display the number of transcoder sessions.
Step 3 Use the show sdspfarm units command to display the number of DSP farms.
Step 4 Obtain the maximum number of transcoder sessions by multiplying the number of transcoder sessions
from Step 2 (configured in Step 1 using the dspfarm transcoder maximum sessions command) by the
number of DSP farms from Step 3.
Note You can unregister all active calls’ transcoding streams with the sdspfarm unregister force command.
Prerequisites
The show interface FastEthernet 0/0 command will yield a MAC address as shown in the following
output:
Router# show interface FastEthernet 0/0
.
.
.
FastEthernet0/0 is up, line protocol is up
Hardware is AmdFE, address is 000a.8aea.ca80 (bia 000a.8aea.ca80)
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. ip source-address ip-address [port port] [any-match | strict-match]
5. sdspfarm units number
6. sdspfarm transcode sessions number
7. sdspfarm tag number device-number
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 ip source-address ip-address [port port] Enables a router to receive messages from Cisco Unified IP
[any-match | strict-match] phones through the router’s IP addresses and ports.
• address—The range is 0 to 5. The default is 0.
Example:
Router(config-telephony)# ip source address
• port port—(Optional) TCP/IP port used for SCCP. The
10.10.10.1 port 3000 default is 2000.
• any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• strict-match—(Optional) Requires strict IP address
checking for registration.
Step 5 sdspfarm units number Specifies the maximum number of DSP farms that are
allowed to be registered to the SCCP server.
Example: • number—The range is 0 to 5. The default is 0.
Router(config-telephony)# sdspfarm units 4
Step 6 sdspfarm transcode sessions number Specifies the maximum number of transcoder sessions for
G.729 allowed by the Cisco Unified CME router.
Example: • One transcoder session consists of two transcoding
Router(config-telephony)# sdspfarm transcode streams between callers using transcode. Use the
sessions 40 maximum number of transcoding sessions and
conference calls that you want your router to support at
one time.
• number—Range is 0 to 128. Default is 0.
Note For the value of number, you can use the value
obtained in step 4 in the “Determining the
Maximum Number of Transcoder Sessions” section
on page 317.
Example:
Router(config-telephony)# end
Configuring the Cisco Unified CME Router to Host a Secure DSP Farm
You must configure the Media Encryption Secure Real-Time Transport Protocol (SRTP) feature on the
Cisco Unified CME router, making it a secure Cisco Unified CME, before it can host a secure DSP farm.
See “Configuring Security” on page 387 for information on configuring a secure Cisco Unified CME.
Registering the DSP Farm with Cisco Unified CME in Secure Mode
The DSP farm can reside on the same router with the Cisco Unified CME or on a different router. Some
of the steps in the following tasks are optional depending the location of the DSP farm.
This section contains the following tasks:
• Obtaining a Digital Certificate from a CA Server, page 320
• Copying the CA Root Certificate of the DSP Farm Router to the Cisco Unified CME Router,
page 326
• Copying the CA Root Certificate of the Cisco Unified CME Router to the DSP farm Router,
page 327
• Configuring Cisco Unified CME to Allow the DSP Farm to Register, page 327
• Verifying DSP Farm Registration with Cisco Unified CME, page 328
Configuring a CA Server
Note Skip this procedure if the DSP farm resides on the same router as the Cisco Unified CME. Proceed to
the “Creating a Trustpoint” section on page 323.
The CA server automatically creates a trustpoint where the certificates are stored. The automatically
created trustpoint stores the CA root certificate.
Prerequisites
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki server label
4. database level complete
5. grant auto
6. database url root-url
7. no shutdown
8. crypto pki trustpoint label
9. revocation-check crl
10. rsakeypair key-label
DETAILED STEPS
Example:
Router# configure terminal
Step 3 crypto pki server label Defines a label for the certificate server and enters
certificate-server configuration mode.
Example: • label—Name for CA certificate server.
Router(config)# crypto pki server dspcert
Example:
Router(cs-server)# exit
Creating a Trustpoint
The trustpoint stores the digital certificate for the DSP farm. To create a trustpoint, perform the following
procedure:
Prerequisites
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki trustpoint label
4. enrollment url ca-url
5. serial-number none
6. fqdn none
7. ip-address none
8. subject-name [x.500-name]
9. revocation-check none
10. rsakeypair key-label
DETAILED STEPS
Example:
Router# configure terminal
Step 3 crypto pki trustpoint label Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA.
Router(config)# crypto pki trustpoint dspcert
Step 4 enrollment url ca-url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • ca-url—URL of the router on which the root CA is
Router(ca-trustpoint)# enrollment url installed.
http://10.3.105.40:80
Step 5 serial-number none Specifies whether the router serial number should be
included in the certificate request.
Example: • none—Specifies that a serial number will not be
Router(ca-trustpoint)# serial-number none included in the certificate request.
Step 6 fqdn none Specifies a fully qualified domain name (FQDN) that will
be included as “unstructuredName” in the certificate
request.
Example:
Router(ca-trustpoint)# fqdn none • none—Router FQDN will not be included in the
certificate request.
Step 7 ip-address none Specifies a dotted IP address or an interface that will be
included as “unstructuredAddress” in the certificate
request.
Example:
Router(ca-trustpoint)# ip-address none • none—Specifies that an IP address is not to be included
in the certificate request.
Step 8 subject-name [x.500-name] Specifies the subject name in the certificate request.
Note The example shows how to format the certificate
Example: subject name to be similar to that of an IP phone’s.
Router(ca-trustpoint)# subject-name cn=vg224,
ou=ABU, o=Cisco Systems Inc.
Prerequisites
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki authenticate trustpoint-label
4. crypto pki enroll trustpoint-label
DETAILED STEPS
Example:
Router# configure terminal
Copying the CA Root Certificate of the DSP Farm Router to the Cisco Unified CME Router
The DSP farm router and Cisco Unified CME router exchanges certificates during the registration
process. These certificates are digitally signed by the CA server of the respective router. For the routers
to accept each others digital certificate, they should have the CA root certificate of each other. Manually
copy the CA root certificate of the DSP farm and Cisco Unified CME router to each other.
Prerequisites
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki trustpoint name
4. enrollment terminal
5. crypto pki export trustpoint pem terminal
6. crypto pki authenticate trustpoint-label
7. You will be prompted to enter the CA certificate. Cut and paste the base 64 encoded certificate at
the command line, then press Enter, and type “quit.” The router prompts you to accept the certificate.
Enter “yes” to accept the certificate.
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(ca-trustpoint)# enrollment terminal
Step 5 crypto pki export trustpoint pem terminal Exports certificates and RSA keys that are associated with
a trustpoint in a privacy-enhanced mail (PEM)-formatted
file.
Example:
Router(ca-trustpoint)# crypto pki export
dspcert pem terminal
Step 6 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate vg224
Note This command is optional if the CA certificate is
already loaded into the configuration.
Step 7 You will be prompted to enter the CA certificate. Cut Completes the copying of the CA root certificate of the DSP
and paste the base 64 encoded certificate at the farm router to the Cisco Unified CME router.
command line, then press Enter, and type “quit.” The
router prompts you to accept the certificate. Enter
“yes” to accept the certificate.
Copying the CA Root Certificate of the Cisco Unified CME Router to the DSP farm Router
Repeat the steps in the “Copying the CA Root Certificate of the DSP Farm Router to the Cisco Unified
CME Router” section on page 326 in the opposite direction, that is, from Cisco Unified CME router to
the DSP farm router.
Prerequisites
Prerequisites
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 sdspfarm units number Specifies the maximum number of digital-signal-processor
(DSP) farms that are allowed to be registered to the Skinny
Client Control Protocol (SCCP) server.
Example:
Router(config-telephony)# sdspfarm units 1
Step 5 sdspfarm transcode sessions number Specifies the maximum number of transcoding sessions
allowed per Cisco Unified CME router.
Example: • number—Declares the number of DSP farm sessions.
Router(config-telephony)# sdspfarm transcode Valid values are numbers from 1 to 128.
sessions 30
Step 6 sdspfarm tag number device-name Permits a DSP farm to register to Cisco Unified CME and
associates it with a SCCP client interface's MAC address.
Example: Note The device-name in this step must be the same as the
Router(config-telephony)# sdspfarm tag 1 vg224 device-name in the associate profile command in
Step 17 of the “Configuring DSP Farms for
NM-HDs and NM-HDV2s” section on page 310.
Step 7 exit Exits telephony-service configuration mode.
Example:
Router(config-telephony)# exit
Prerequisites
SUMMARY STEPS
DETAILED STEPS
Step 1 Use the show sccp [statistics | connections] command to display the SCCP configuration information
and current status.
Router# show sccp statistics
Use the show sccp connections command to display information about the connections controlled by
the SCCP transcoding and conferencing applications. In the following example, the secure value of the
stype field indicates that the connection is encrypted:
Router# show sccp connections
Step 2 Use the show sdspfarm units command to display the configured and registered DSP farms.
Router# show sdspfarm units
Step 3 Use the show sdspfarm sessions command to display the transcoding streams.
Router# show sdspfarm sessions
Stream-ID:1 mtp:1 10.10.10.3 18404 Local:2000 START
usage:Ip-Ip
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:2
Step 4 Use the show sdspfarm sessions summary command to display a summary view the transcoding
streams.
Router# show sdspfarm sessions summary
Step 5 Use the show sdspfarm sessions active command to display the transcoding streams for all active
sessions.
Router# show sdspfarm sessions active
Stream-ID:1 mtp:1 10.10.10.3 18404 Local:2000 START
usage:Ip-Ip
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:2
Step 6 Use the show sccp connections details command to display the SCCP connections details such as
call-leg details.
Router# show sccp connections details
Step 7 Use the debug sccp {all | errors | events | packets | parser} command to set debugging levels for SCCP
and its applications.
Step 8 Use the debug dspfarm {all | errors | events | packets} command to set debugging levels for DSP-farm
service
Step 9 Use the debug ephone mtp command to enable Message Transfer Part (MTP) debugging. Use this debug
command with the debug ephone mtp, debug ephone register, debug ephone state, and debug ephone
pak commands.
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 sccp ip precedence value (Optional) Sets the IP precedence value to increase the
priority of voice packets over connections controlled by
SCCP.
Example:
Router(config)# sccp ip precedence 5
Step 4 dspfarm rtp timeout seconds (Optional) Configures the Real-Time Transport Protocol
(RTP) timeout interval if the error condition “RTP port
unreachable” occurs.
Example:
Router(config)# dspfarm rtp timeout 60
Step 5 dspfarm connection interval seconds (Optional) Specifies how long to monitor RTP inactivity
before deleting an RTP stream.
Example:
Router(config)# dspfarm connection interval 60
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config)# end
telephony-service
ip source-address 10.5.49.200 port 2000
sdspfarm units 4
sdspfarm transcode sessions 40
sdspfarm tag 1 mtp000a8eaca80
sdspfarm tag 2 mtp123445672012
telephony-service
ip source-address 10.5.49.200 port 2000
sdspfarm units 1
sdspfarm transcode sessions 40
sdspfarm tag 1 mtp000a8eaca80
sdspfarm tag 2 mtp123445672012
Where to go Next
Music on Hold
Music on hold can require transcoding resources. See “Configuring Music on Hold” on page 771.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 17 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the Cisco Unified Communications Manager Express (Cisco Unified CME)
graphical user interface (GUI) and explains how to set it up for three different levels of user.
Contents
• Prerequisites for Enabling the GUI, page 339
• Restrictions for Enabling the GUI, page 340
• Information About Enabling the GUI, page 340
• How to Enable the GUI, page 341
• Configuration Examples for Enabling the GUI, page 350
• Additional References, page 352
• Feature Information for Enabling the GUI, page 354
AAA Authentication
The GUI supports authentication, authorization, and accounting (AAA) authentication for system
administrators through a remote server when this capability is enabled with the ip http authentication
command. If authentication through the server fails, the local router is searched.
Using the ip http authentication command prevents unauthorized users from accessing the
Cisco Unified CME router. If this command is not used, the enable password for the router is the only
requirement to authenticate user access to the GUI. Instead, we recommend you use the local or
TACACS authentication options, configured as part of a global AAA framework. By explicitly using the
ip http authentication command, you designate alternative authentication methods, such as by a local
login account or by the method that is specified in the AAA configuration on the Cisco Unified CME
router. If you select the AAA authentication method, you must also define an authentication method in
your AAA configuration.
For information on configuring AAA authentication, see the “Configuring Authentication” chapter of
the Cisco IOS Security Configuration Guide for your Cisco IOS release.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip http server
4. ip http path flash:
5. ip http authentication {aaa | enable | local | tacacs}
6. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ip http server Enables the HTTP server on the Cisco Unified CME
router.
Example:
Router(config)# ip http server
Step 4 ip http path flash: Sets the location of the HTML files used by the HTTP
web server to flash memory on the router.
Example:
Router(config)# ip http path flash:
Step 5 ip http authentication {aaa | enable | local | Specifies the method of authentication for the HTTP
tacacs} server. Default is the enable keyword.
• aaa—Indicates that the authentication method
Example: used for the AAA login service should be used
Router(config)# ip http authentication aaa for authentication. The AAA login service
method is specified by the aaa authentication
login command.
• enable—Uses the enable password. This is the
default if this command is not used.
• local—Uses login username, password, and
privilege level access combination specified in
the local system configuration (by the username
command).
• tacacs—Uses TACACS (or XTACACS) server.
Step 6 exit Returns to privileged EXEC mode.
Example:
Router(config)# exit
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. web admin system name username {password string | secret {0 | 5} string}
5. dn-webedit
6. time-webedit
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 web admin system name username {password string | Defines username and password for a system
secret {0 | 5} string} administrator.
• name username—System administrator
Example: username. Default is Admin.
Router(config-telephony)# web admin system name pwa3
secret 0 wp78pw • password string—String to verify system
administrator’s identity. Default is empty string.
• secret {0 | 5} string—Digit specifies state of
encryption of the string that follows:
– 0—Password that follows is not encrypted.
– 5—Password that follows is encrypted using
Message Digest 5 (MD5).
Note The secret 5 keyword pair is used in the
output of show commands when encrypted
passwords are displayed. It indicates that the
password that follows is encrypted.
Example:
Router(config-telephony)# end
Restrictions
The Cisco Unified CME GUI requires Microsoft Internet Explorer 5.5 or a later version. Other browsers
are not supported.
DETAILED STEPS
where router_ipaddress is the IP address of your Cisco Unified CME router. For example, if the IP
address of your Cisco Unified CME router is 10.10.10.176, enter the following:
http://10.10.10.176/ccme.html
The Cisco Unified CME system evaluates your privilege level and presents the appropriate window.
Note that users with Cisco IOS software privilege level 15 also have system-administrator-level
privileges in the Cisco Unified CME GUI after being authenticated locally or remotely through AAA.
The ip http authentication command that is configured on the Cisco Unified CME router determines
where authentication occurs.
Step 3 After you login and are authenticated, the system displays one of the following home pages, based on
your user level:
• The system administrator home page.
• The customer administrator sees a reduced version of the options available on the system
administrator page, according to the XML configuration file that the system administrator created.
• The phone user home page.
After you log in successfully, online help is available from the Help menu.
SUMMARY STEPS
DETAILED STEPS
Step 1 Copy the XML template that you downloaded from the Cisco Software Center and open it in any text
editor (see the “XML Configuration File Template: Example” section on page 350). Give the file a name
that is meaningful to you and that uses “xml” as its suffix. For example, you could name the file
“custadm.xml.”
Step 2 Edit the XML template. Within the template, each line that starts with a title enclosed in angle brackets
describes an XML object and matches an entity name in the Cisco CME GUI. For example,
“<AddExtension>” refers to the Add Extension capability, and “<Type>” refers to the Type field on the
Add Extension window. For each object in the template, you have a choice of actions. Your choices
appear within brackets; for example, “[Hide | Show]” indicates that you have a choice between whether
this object is hidden or visible when a customer administrator logs in to the GUI. Delete the action that
you do not want and the vertical bar and brackets around the actions.
For example, to hide the Sequence Number field, change the following text in the template file:
<SequenceNumber> [Hide | Show] </SequenceNumber>
Edit every line in the template until you have changed each choice in brackets to a single action and you
have removed the vertical bars and brackets. A sample XML file is shown in the “XML Configuration
File: Example” section on page 351.
Step 3 Copy the file to a TFTP or FTP server that can be accessed by the Cisco Unified CME router.
Step 4 Copy your file to flash memory on the Cisco Unified CME router.
Router# copy tftp flash
Prerequisites
• Enable a system administrator account for GUI access. See the “Enabling GUI Access for the
System Administrator” section on page 343.
• Create the XML configuration file for the customer administrator GUI. See the “Creating a
Customized XML File for Customer Administrator GUI” section on page 345.
• Reload the XML file using the web customize load command if you have made changes to the
customer administrator GUI.
Using the Cisco Unified CME GUI to Define a Customer Administrator Account
To allow the system administrator to use the GUI to create a customer administrator account, perform
the following steps.
DETAILED STEPS
Step 1 From the Configure System Parameters menu, choose Administrator’s Login Account.
Step 2 Complete the Admin User Name (username), Admin User Type (Customer), and New Password fields
for the user that you are defining as a customer administrator. Type the password again to confirm it.
Step 3 Click Change for your changes to become effective.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. web admin customer name username password string
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
Prerequisites
• Enable a system administrator account for GUI access. See the “Enabling GUI Access for the
System Administrator” section on page 343.
Using the Cisco Unified CME GUI to Define a Phone User Account
To create a phone user account by using the Cisco Unified CME GUI, perform the following steps.
DETAILED STEPS
Step 1 From the Configure Phones menu, choose Add Phone to add GUI access for a user with a new phone or
Change Phone to add GUI access for a user with an existing phone. The Add Phone screen or the
Change Phone screen displays.
Step 2 Enter a username and password in the Login Account area of the screen. If you are adding a new phone,
complete the other fields as appropriate.
Step 3 Click Change for your edits to become effective.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. username username password password
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 2
Step 4 username username password password Assigns a phone user login account name and
password.
Example: • This allows the phone user to log in to the
Router(config-ephone)# username prx password pk59wq Cisco Unified CME GUI to change a limited
number of personal settings.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Step 1 Verify you are using Microsoft Internet Explorer 5.5 or a later version. No other browser is supported.
Step 2 Clear your browser cache or history.
Step 3 Verify that the GUI files in router flash memory are the correct version for the version of
Cisco Unified CME that you have. Compare the filenames in flash memory with the list in the
Cisco Unified CME software archive that you downloaded. Compare the sizes of files in flash memory
with the sizes of the files in the tar archive called cme-3.2.0-gui.tar (or a later version of the file) to
ensure that you have the most recent files installed in flash memory. The latest version can be
downloaded from the Cisco Unified CME Software Download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.
telephony-service
web admin system name pwa3 secret 0 wp78pw
web admin customer name user44 password pw10293847
dn-webedit
time-webedit
ephone 25
username prx password pswd
<Extension>
<!-- Control both view and change, and possible add or delete -->
<SequenceNumber> [Hide | Show] </SequenceNumber>
<Type> [Hide | Show] </Type>
<Huntstop> [Hide | Show] </Huntstop>
<Preference> [Hide | Show] </Preference>
<HoldAlert> [Hide | Show] </HoldAlert>
<Phone>
<!-- control both view and change, and possible add and delete --->
<SequenceNumber> [Hide | Show] </SequenceNumber>
</Phone>
<System>
<!-- Control View Only -->
<PhoneURL> [Hide | Show] </PhoneURL>
<PhoneLoad> [Hide | Show]</PhoneLoad>
<CallHistory> [Hide | Show] </CallHistory>
<MWIServer> [Hide | Show] </MWIServer>
<!-- Control Either View and Change or Change Only -->
<TransferPattern attr=[Both | Change]> [Hide | Show] </TransferPattern>
<VoiceMailNumber attr=[Both | Change]> [Hide | Show] </VoiceMailNumber>
<MaxNumberPhone attr=[Both | Change]> [Hide | Show] </MaxNumberPhone>
<DialplanPattern attr=[Both | Change]> [Hide | Show] </DialplanPattern>
<SecDialTone attr=[Both | Change]> [Hide | Show] </SecDialTone>
<Timeouts attr=[Both | Change]> [Hide | Show] </Timeouts>
<CIDBlock attr=[Both | Change]> [Hide | Show] </CIDBlock>
<HuntGroup attr=[Both | Change]> [Hide | Show] </HuntGroup>
<NightSerBell attr=[Both | Change]> [Hide | Show] </NightSerBell>
<!-- Control Change Only -->
<!-- Take Higher Precedence over CLI "time-web-edit" -->
<Time> [Hide | Show] </Time>
</System>
<Function>
<AddLineToPhone> [No | Yes] </AddLineToPhone>
<DeleteLineFromPhone> [No | Yes] </DeleteLineFromPhone>
<NewDnDpCheck> [No | Yes] </NewDnDpCheck>
<MaxLinePerPhone> [1-6] </MaxLinePerPhone>
</Function>
</Presentation>
<Extension>
<SequenceNumber> Hide </SequenceNumber>
<Type> Hide </Type>
<Huntstop> Hide </Huntstop>
<Preference> Hide </Preference>
<Phone>
<SequenceNumber> Hide </SequenceNumber>
</Phone>
<System>
<PhoneURL> Hide </PhoneURL>
<PhoneLoad> Hide </PhoneLoad>
<CallHistory> Hide </CallHistory>
<MWIServer> Hide </MWIServer>
<TransferPattern attr=Both> Hide </TransferPattern>
<VoiceMailNumber attr=Both> Hide </VoiceMailNumber>
<MaxNumberPhone attr=Both> Hide </MaxNumberPhone>
<DialplanPattern attr=Change> Hide </DialplanPattern>
<SecDialTone attr=Both> Hide </SecDialTone>
<Timeouts attr=Both> Hide </Timeouts>
<CIDBlock attr=Both> Hide </CIDBlock>
<HuntGroup attr=Change> Hide </HuntGroup>
<NightSerBell attr=Change> Hide </NightSerBell>
<Time> Hide </Time>
</System>
<Function>
<AddLineToPhone> No </AddLineToPhone>
<DeleteLineFromPhone> No </DeleteLineFromPhone>
<MaxLinePerPhone> 4 </MaxLinePerPhone>
</Function>
</Presentation>
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Security Command Reference
• Cisco IOS Software Releases 12.4T Command References
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note The following table lists the Cisco Unified CME version that introduced support for a given feature.
Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes how to integrate your voice-mail system with Cisco Unified Communications
Manager Express (Cisco Unified CME).
Contents
• Prerequisites, page 355
• Information About Voice-Mail Integration, page 355
• How to Configure Voice-Mail Integration, page 360
• Configuration Examples for Voice-Mail Integration, page 381
• Additional References, page 385
• Feature Information for Voice-Mail Integration, page 386
Prerequisites
• Voice mail must be installed and configured on your network.
• Calls can be successfully completed between phones on the same Cisco Unified CME router.
Note Cisco Unified CME and Cisco Unity Express must both be configured before they can be integrated.
Figure 17 SIP MWI to ISDN QSIG When Voice Mail and Cisco Router are On the Same LAN
U
146430
LAN
It makes no difference if the SIP Unsolicited NOTIFY is received via LAN or WAN as long as the PBX
is connected to the Cisco router, and not to the remote voice mail server.
In Figure 18, a voice mail server and Cisco Unified CME are connected to the same LAN and a remote
Cisco Unified CME is connected across the WAN. In this scenario, the protocol translation is performed
at the remote Cisco router and the QSIG MWI message is sent to the PBX.
Figure 18 SIP MWI to ISDN QSIG When PBX is Connected to a Remote Cisco Router
WAN
146570
PBX
Cisco Unified CME Cisco Unified CME
LAN
Note The same telephone number is configured for voice messaging for all SCCP phones in
Cisco Unified CME.
Prerequisites
• Voicemail phone number must be a valid number; directory number and number for voicemail
phone number must be configured. For configuration information, see “Configuring Phones to Make
Basic Calls” on page 147.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. voicemail phone-number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters voice register global configuration mode to set
parameters for all supported phones in Cisco Unified CME.
Example:
Router(config)# telephony-service
Step 4 voicemail phone-number Defines the telephone number that is speed-dialed when the
Messages button on a Cisco Unified IP phone is pressed.
Example: • phone-number—Same phone number is configured for
Router(config-telephony)# voice mail 0123 voice messaging for all SCCP phones in a
Cisco Unified CME.
Example:
Router(config-telephony)# end
What to Do Next
• (Cisco Unified CME 4.0 or a later version only) To set up a mailbox selection policy, see the
“SCCP: Configuring a Mailbox Selection Policy” section on page 362.
• To set up DTMF integration patterns for connecting to analog voice-mail applications, see the
“Enabling DTMF Integration for Legacy Voice-Mail Applications” section on page 367.
• To connect to a remote SIP-based IVR or Cisco Unity, or to connect to a remote SIP-PSTN that goes
through the PSTN to a voice-mail or IVR application, see the “Enabling DTMF Integration Using
RFC 2833” section on page 369.
• To connect to a Cisco Unity Express system, configure a nonstandard SIP NOTIFY format. See the
“Enabling DTMF Integration Using SIP NOTIFY” section on page 372.
SCCP: Setting Mailbox Selection Policy for Cisco Unity Express or PBX Voice-Mail Number
To set a policy for selecting a mailbox for calls from a Cisco Unified CME system that are diverted
before being sent to a Cisco Unity Express or PBX voice-mail pilot number, perform the following steps.
Prerequisites
Restrictions
In the following scenarios, the mailbox selection policy can fail to work properly:
• The last redirecting endpoint is not hosted on Cisco Unified CME. This may rarely occur with a
PBX.
• A call is forwarded across several SIP trunks. Multiple SIP Diversion Headers (stacking hierarchy)
are not supported in Cisco IOS software.
• A call is forwarded across non-Cisco voice gateways that do not support the optional H450.3
originalCalledNr field.
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
or
dial-peer voice tag pots
4. mailbox-selection [last-redirect-num | orig-called-num]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.
or
• tag—Identifies the dial peer. Valid entries are from 1 to
dial-peer voice tag pots 2147483647.
Note This command should be used on the outbound dial
Example: peer associated with the pilot number of the
Router(config)# dial-peer voice 7000 voip voice-mail system. For systems using
or Cisco Unity Express, this is a VoIP dial peer. For
Router(config)# dial-peer voice 35 pots systems using PBX-based voice mail, this is a POTS
dial peer.
Step 4 mailbox-selection [last-redirect-num | Sets a policy for selecting a mailbox for calls that are
orig-called-num] diverted before being sent to a voice-mail line.
• last-redirect-num—(PBX voice mail only) The
Example: mailbox number to which the call will be sent is the last
Router(config-dial-peer)# mailbox-selection number to divert the call (the number that sends the call
orig-called-num
to the voice-mail pilot number).
• orig-called-num—(Cisco Unity Express only) The
mailbox number to which the call will be sent is the
number that was originally dialed before the call was
diverted.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
What to Do Next.
• To use voice mail on a SIP network that connects to a Cisco Unity Express system, configure a
nonstandard SIP NOTIFY format. See the “Enabling DTMF Integration Using SIP NOTIFY”
section on page 372.
Prerequisites
Restrictions
This feature might not work properly in certain network topologies, including the following cases:
• When the last redirecting endpoint is not hosted on Cisco Unified CME. This may rarely occur with
a PBX.
• When a call is forwarded across several SIP trunks. Multiple SIP Diversion Headers (stacking
hierarchy) are not supported in Cisco IOS software.
• When a call is forwarded across other voice gateways that do not support the optional H450.3
originalCalledNr field.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. mailbox-selection last-redirect-num
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 exit Exits dial-peer configuration mode.
Example:
Router(config-dial-peer)# exit
Example:
Router(config)# ephone-dn 752
Step 5 mailbox-selection [last-redirect-num] Sets a policy for selecting a mailbox for calls that are
diverted before being sent to a Cisco Unity voice-mail pilot
number.
Example:
Router(config-ephone-dn)# mailbox-selection
last-redirect-num
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
What to Do Next
• To use a remote SIP-based IVR or Cisco Unity, or to connect Cisco Unified CME to a remote
SIP-PSTN that goes through the PSTN to a voice-mail or IVR application, see the “Enabling DTMF
Integration Using RFC 2833” section on page 369.
Note The same telephone number is configured for voice messaging for all SIP phones in Cisco Unified CME.
The call forward b2bua command enables call forwarding and designates that calls that are forwarded
to a busy or no-answer extension be sent to a voicemail box.
Prerequisites
• Directory number and number for voicemail phone number must be configured. For configuration
information, see “Configuring Phones to Make Basic Calls” on page 147.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. voicemail phone-number
5. exit
6. voice register dn dn-tag
7. call-forward b2bua busy directory-number
8. call-forward b2bua mailbox directory-number
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 voicemail phone-number Defines the telephone number that is speed-dialed when the
Messages button on a Cisco Unified IP phone is pressed.
Example: • phone-number—Same phone number is configured for
Router(config-register-global)# voice mail 1111 voice messaging for all SIP phones in a
Cisco Unified CME.
Step 5 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-global)# exit
Step 6 voice register dn dn-tag Enters voice register dn mode to define a directory number
for a SIP phone, intercom line, voice port, or an MWI.
Example:
Router(config-register-global)# voice register
dn 2
Step 7 call-forward b2bua busy directory-number Enables call forwarding for a SIP back-to-back user agent
so that incoming calls to an extension that is busy will be
forwarded to the designated directory number.
Example:
Router(config-register-dn)# call-forward b2bua
busy 1000
Step 8 call-forward b2bua mailbox directory-number Designates voice mailbox to use at the end of a chain of call
forwards. Incoming calls have been forwarded to a busy or
no-answer extension will be forwarded to the
Example:
Router(config-register-dn)# call-forward b2bua
directory-number specified.
mailbox 2200
What to Do Next
• To set up DTMF integration patterns for connecting to analog voice-mail applications, see the
“Enabling DTMF Integration for Legacy Voice-Mail Applications” section on page 367.
• To use a remote SIP-based IVR or Cisco Unity, or to connect to a remote SIP-PSTN that goes
through the PSTN to a voice-mail or IVR application, see the “Enabling DTMF Integration Using
RFC 2833” section on page 369.
• To connect to a Cisco Unity Express system, configure a nonstandard SIP NOTIFY format, see the
“Enabling DTMF Integration Using SIP NOTIFY” section on page 372.
Note You can configure multiple tags and tokens for each pattern, depending on the voice-mail system and
type of access.
SUMMARY STEPS
1. enable
2. configure terminal
3. vm-integration
4. pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
5. pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
6. pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
7. pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
8. pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 vm-integration Enters voice-mail integration configuration mode and
enables voice-mail integration with DTMF and an analog
voice-mail system.
Example:
Router(config) vm-integration
Step 4 pattern direct tag1 {CGN | CDN | FDN} [tag2 Configures the DTMF digit pattern forwarding necessary to
{CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] activate the voice-mail system when the user presses the
[last-tag]
messages button on the phone.
• The tag attribute is an alphanumeric string fewer than
Example: four DTMF digits in length. The alphanumeric string
Router(config-vm-integration) pattern direct
consists of a combination of four letters (A, B, C, and D),
2 CGN *
two symbols (* and #), and ten digits (0 to 9). The tag
numbers match the numbers defined in the voice-mail
system’s integration file, immediately preceding either
the number of the calling party, the number of the called
party, or a forwarding number.
• The keywords—CGN, CDN, and FDN—configure the
type of call information sent to the voice-mail system,
such as calling number (CGN), called number (CDN), or
forwarding number (FDN).
Example:
Router(config-vm-integration)# exit
What to Do Next
After integrating configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI)
notification for either the MWI outcall, unsolicited notify, or subscribe/notify mechanism. See the
“SCCP: Configuring a Phone Line for MWI Outcall” section on page 374.
• When SIP is used to connect Cisco Unified CME to a remote SIP-PSTN voice gateway that goes
through the PSTN to a voice-mail or IVR application.
Note If the T38 Fax Relay feature is also configured on this IP network, we recommend that you either
configure the voice gateways to use a payload type other than PT96 or PT97 for fax relay negotiation,
or depending on whether the SIP endpoints support different payload types, configure Cisco Unified
CME to use a payload type other than PT96 or PT97 for DTMF.
Prerequisites
• Configure the codec or voice-class codec command for transcoding between G.711 and G.729. See
“Configuring Phones to Make Basic Calls” on page 147.
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
4. description string
5. destination-pattern string
6. session protocol sipv2
7. session target {dns:address | ipv4:destination-address}
8. dtmf-relay rtp-nte
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode to define a VoIP dial
peer for the voice-mail system.
Example: • tag—Defines the dial peer being configured. Range is
Router (config)# dial-peer voice 123 voip from 1 to 2147483647.
Step 4 description string (Optional) Associates a description with the dial peer being
configured. Enter a string of up to 64 characters.
Example:
Router (config-voice-dial-peer)# description CU
pilot
What to Do Next
After integrating configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI)
notification for either the MWI outcall, unsolicited notify, or subscribe/notify mechanism. See the
“SCCP: Configuring a Phone Line for MWI Outcall” section on page 374.
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
4. description string
5. destination-pattern string
6. b2bua
7. session protocol sipv2
8. session target {dns:address | ipv4:destination-address}
9. dtmf-relay sip-notify
10. codec g711ulaw
11. no vad
12. end
DETAILED STEPS
Example:
Router# configure terminal#
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode to define a VoIP dial
peer for the voice-mail system.
Example: • tag—Defines the dial peer being configured. Range is
Router (config)# dial-peer voice 2 voip from 1 to 2147483647.
Step 4 description string (Optional) Associates a description with the dial peer being
configured. Enter a string of up to 64 characters.
Example:
Router (config-voice-dial-peer)# description
cue pilot
Step 5 destination-pattern string Specifies the pattern of the numbers that the user must dial
to place a call.
Example: • string—Prefix or full E.164 number.
Router (config-voice-dial-peer)#
destination-pattern 20
What to Do Next
After configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI). See the
“SCCP: Configuring a Phone Line for MWI Outcall” section on page 374.
Prerequisites
• Directory number and number for MWI line must be configured. For configuration information, see
“Configuring Phones to Make Basic Calls” on page 147.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mwi-line line-number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 36
Step 4 mwi-line line-number Selects a phone line to be monitored for voice-mail messages.
• line-number—Number of phone line to receive MWI
Example: notification. Range: 1 to 34. Default: 1.
Router(config-ephone)# mwi-line 3
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Prerequisites
• Directory number and number for MWI line must be configured. For configuration information, see
“Configuring Phones to Make Basic Calls” on page 147.
Restrictions
• Audible MWI is supported only in Cisco Unified CME 4.0(2) and later versions.
• Audible MWI is supported only on Cisco Unified IP Phone 7931G and Cisco Unified IP Phone
7911.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mwi-line line-number
5. ephone-dn dn-tag
6. mwi-type {visual | audio | both}
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 36
Step 4 mwi-line line-number Selects a phone line to receive MWI treatment.
• line-number—Number of phone line to receive MWI
Example: notification. Range: 1 to 34. Default: 1.
Router(config-ephone)# mwi-line 3
Example:
Router(config-ephone)# end
Prerequisites
• Cisco CME 3.4 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. mwi reg-e164
5. mwi stutter
6. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-register-global)# end
Prerequisites
Restrictions
• For Cisco Unified CME 4.1 and later versions, the Call Forward All, Presence, and
Message-Waiting Indication (MWI) features require that SIP phones must be configured with a
directory number by using the number command with the dn keyword; direct line numbers are not
supported.
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1
Step 4 mwi Enables a specific directory number to receive MWI
notification.
Example:
Router(config-register-dn)# mwi
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone-dn)# end
Note We recommend using the Subscribe/NOTIFY method rather than an Unsolicited NOTIFY when
possible.
Prerequisites
Restrictions
• For Cisco Unified CME 4.1 and later versions, the Call Forward All, Presence, and
Message-Waiting Indication (MWI) features require that SIP phones must be configured with a
directory number by using the number command with the dn keyword; direct line numbers are not
supported.
• The SIP MWI - QSIG Translation feature in Cisco Unified CME 4.1 does not support Subscribe
NOTIFY.
• Cisco Unified IP Phone 7960, 7940, 7905, and 7911 support only Unsolicited NOTIFY for MWI.
SUMMARY STEPS
1. enable
2. configure terminal
3. sip-ua
4. mwi-server {ipv4:destination-address | dns:host-name} [unsolicited]
5. exit
6. voice register dn dn-tag
7. mwi
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 sip-ua Enters Session Initiation Protocol (SIP) user agent (ua)
configuration mode for configuring the user agent.
Example:
Router(config)# sip-ua
Step 4 mwi-server {ipv4:destination-address | Specifies voice-mail server settings on a voice gateway or
dns:host-name} [unsolicited] UA.
Note The sip-server and mwi expires commands under
Example: the telephony-service configuration mode have
Router(config-sip-ua)# mwi-server been migrated to mwi-server to support DNS
ipv4:1.5.49.200
format of the SIP server.
or
Router(config-sip-ua)# mwi-server
dns:server.yourcompany.com unsolicited
Prerequisites
• Cisco Unified CME 4.0 or a later version.
• Directory number for receiving MWI Unsolicited NOTIFY must be configured. For information, see
“SIP: Configuring a Directory Number for MWI NOTIFY” section on page 378.
SUMMARY STEPS
1. enable
2. telephony-service
3. mwi prefix prefix-string
4. end
DETAILED STEPS
Example:
Router(config)# telephony-service
Step 3 mwi prefix prefix-string Specifies a string of digits that, if present before a known
Cisco Unified CME extension number, should be
recognized as a prefix.
Example:
Router(config-telephony)# mwi prefix 555 • prefix-string—Digit string. The maximum prefix length
is 32 digits.
Step 4 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
The following example sets a policy to select the mailbox of the last number that the call was diverted
to before being diverted to a Cisco Unity voice-mail system with the pilot number 8000.
ephone-dn 825
number 8000
mailbox-selection last-redirect-num
ephone-dn 21
number 2021
ephone-dn 22
number 2022
ephone-dn 23
number 2023
ephone-dn 24
number 2024
ephone-dn 25
number 2025
ephone 18
button 1:20 2o21,22,23,24,25 3x2 5:26
mwi-line 2
The following example enables MWI on ephone 17 for line 3 (extension 609). In this example, the button
numbers do not match the line numbers because buttons 2 and 4 are not used. The line numbers in this
example are as follows:
• Line 1—Button 1—Extension 607
• Button 2—Unused
• Line 2—Button 3—Extension 608
• Button 4—Unused
• Line 3—Button 5—Extension 609
ephone-dn 17
number 607
ephone-dn 18
number 608
ephone-dn 19
number 609
ephone 25
button 1:17 3:18 5:19
mwi-line 3
telephony-service
mwi prefix 555
voice register dn 1
number 1234
mwi
voice register dn 1
number 1234
mwi
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 19 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the phone authentication support in Cisco Unified Communications Manager
Express (Cisco Unified CME) and the Media Encryption (SRTP) on Cisco Unified CME feature which
provide the following secure voice call capabilities:
• Secure call control signaling and media streams in Cisco Unified CME networks using Secure
Real-Time Transport Protocol (SRTP) and H.323 protocols.
• Secure supplementary services for Cisco Unified CME networks using H.323 trunks.
• Secure Cisco VG224 Analog Phone Gateway endpoints.
Contents
• Prerequisites for Security, page 388
• Restrictions for Security, page 388
• Information About Security, page 389
• How to Configure Security, page 402
• Configuration Examples for Security, page 438
• Where to Go Next, page 458
• Additional References, page 459
• Feature Information for Security, page 460
Media Encryption
• Cisco Unity 4.2 or later version
• Cisco IOS Release 12.5 or a later release on the Cisco VG224 Analog Phone Gateway
• Cisco Unified CME 4.2 or a later version
Media Encryption
• Secure three-way software conference is not supported. A secure call beginning with SRTP will
always fall back to nonsecure Real-Time Transport Protocol (RTP) when it is joined to a conference.
• If a party drops from a three-party conference, the call between the remaining two parties returns to
secure if the two parties are SRTP-capable local Skinny Client Control Protocol (SCCP) endpoints
to a single Cisco Unified CME and the conference creator is one of the remaining parties. If either
of the two remaining parties are only RTP-capable, the call remains nonsecure. If the two remaining
parties are connected through FXS, PSTN, or VoIP, the call remains nonsecure.
• Calls to Cisco Unity Express are not secure.
• Music on Hold (MOH) is not secure.
• Video calls are not secure.
• Modem relay and T.3 fax relay calls are not secure.
• Media flow-around is not supported for call transfer and call forward.
• Conversion between inband tone and RFC 2833 DTMF is not supported. RFC 2833 DTMF handling
is supported when encryption keys are sent to secure DSP farm devices but is not supported for
codec passthrough.
• Secure Cisco Unified CME does not support SIP trunks; only H.323 trunks are supported.
Table 20 lists supported gateways, network modules, and codecs for Media Encryption (SRTP) on
Cisco Unified CME.
Table 20 Supported Gateways, Network Modules, and IP Phones for Media Encryption (SRTP) on
Cisco Unified CME
Phone Authentication
• Phone Authentication Overview, page 390
• Public Key Infrastructure, page 391
• Phone Authentication Components, page 391
• Phone Authentication Process, page 394
• Startup Messages, page 395
• Configuration File Maintenance, page 395
• CTL File Maintenance, page 396
• CTL Client and Provider, page 396
• Manually Importing MIC Root Certificate, page 397
Media Encryption
• Feature Design of Media Encryption, page 397
• Secure Cisco Unified CME, page 398
• Secure Supplementary Services, page 399
• Secure Transcoding for Remote Phones with DSP Farm Transcoding Configured, page 400
• Secure Cisco Unified CME with Cisco Unity Express, page 401
• Secure Cisco Unified CME with Cisco Unity, page 401
Phone Authentication
The phone authentication process occurs between the Cisco Unified CME router and a supported device
when each entity accepts the certificate of the other entity; only then does a secure connection between
the entities occur. Phone authentication relies on the creation of a Certificate Trust List (CTL) file, which
is a list of known, trusted certificates and tokens. Phones communicate with Cisco Unified CME using
a secure transport-layer-session (TLS) connection, which requires that the following criteria be met:
• A certificate must exist on the phone.
• A phone configuration file must exist on the phone, and the Cisco Unified CME entry and certificate
must exist in the file.
File Authentication
The file authentication process validates digitally signed files that a phone downloads from a Trivial File
Transfer Protocol (TFTP) server—for example, configuration files, ring list files, locale files, and CTL
files. When the phone receives these types of files from the TFTP server, the phone validates the file
signatures to verify that file tampering did not occur after the files were created.
Signaling Authentication
The signaling authentication process, also known as signaling integrity, uses the TLS protocol to
validate that signaling packets have not been tampered with during transmission. Signaling
authentication relies on the creation of the CTL file.
Component Definition
certificate An electronic document that binds a user's or device's name to its
public key. Certificates are commonly used to validate digital
signatures. Certificates are needed for authentication during secure
communication. An entity obtains a certificate by enrolling with the
CA.
signature An assurance from an entity that the transaction it accompanies is
authentic. The entity’s private key is used to sign transactions and the
corresponding public key is used for decryption.
Component Definition
RSA key pair RSA is a public key cryptographic system developed by Ron Rivest,
Adi Shamir, and Leonard Adleman.
An RSA key pair consists of a public key and a private key. The public
key is included in a certificate so that peers can use it to encrypt data
that is sent to the router. The private key is kept on the router and used
both to decrypt the data sent by peers and to digitally sign transactions
when negotiating with peers.
You can configure multiple RSA key pairs to match policy
requirements, such as key length, key lifetime, and type of keys, for
different certificate authorities or for different certificates.
certificate server A certificate server generates and issues certificates on receipt of
trustpoint legitimate requests. A trustpoint with the same name as the certificate
server stores the certificates. Each trustpoint has one certificate plus
a copy of the CA certificate.
certification authority (CA) The root certificate server. It is responsible for managing certificate
requests and issuing certificates to participating network devices.
This service provides centralized key management for participating
devices and is explicitly trusted by the receiver to validate identities
and to create digital certificates. The CA can be a Cisco IOS CA on
the Cisco Unified CME router, a Cisco IOS CA on another router, or
a third-party CA.
registration authority (RA) Records or verifies some or all of the data required for the CA to issue
certificates. It is required when the CA is a third-party CA or
Cisco IOS CA is not on the Cisco Unified CME router.
certificate trust list (CTL) file A mandatory structure that contains the public key information
(server identities) of all the servers with which the IP phone needs to
CTL client
interact (for example, the Cisco Unified CME server, TFTP server,
CTL provider and CAPF server). The CTL file is digitally signed by the system
administrator security token (SAST).
After you configure the CTL client, it creates the CTL file and makes
it available in the TFTP directory. The CTL file is signed using the
SAST certificate’s corresponding private key. An IP phone is then
able to download this CTL file from the TFTP directory. The filename
format for each phone’s CTL file is CTLSEP<mac-addr>.tlv.
When the CTL client is run on a router in the network that is not a
Cisco Unified CME router, you must configure a CTL provider on
each Cisco Unified CME router in the network. Similarly, if a CTL
client is running on one of two Cisco Unified CME routers in a
network, a CTL provider must be configured on the other
Cisco Unified CME router. The CTL protocol transfers information to
and from the CTL provider that allows the second Cisco Unified CME
router to be trusted by phones and vice versa.
certificate revocation list File that contains certificate expiration dates and used to determine
(CRL) whether a certificate that is presented is valid or revoked.
Component Definition
system administrator security Part of the CTL client that is responsible for signing the CTL file. The
token (SAST) Cisco Unified CME certificate and its associated key pair are used for
the SAST function. There are actually two SAST records pertaining
to two different certificates in the CTL file for security reasons. They
are known as SAST1 and SAST2. If one of the certificates is lost or
compromised, then the CTL client regenerates the CTL file using the
other certificate. When a phone downloads the new CTL file, it
verifies with only one of the two original public keys that was
installed earlier. This mechanism is to prevent IP phones from
accepting CTL files from unknown sources.
certificate authority proxy Entity that issues certificates (LSCs) to phones that request them. The
function (CAPF) CAPF is a proxy for the phones, which are unable to directly
communicate with the CA. The CAPF can also perform the following
certificate-management tasks:
• Upgrade existing locally significant certificates on the phones.
• Retrieve phone certificates for viewing and troubleshooting.
• Delete locally significant certificates on the phone.
manufacture-installed Phones need certificates to engage in secure communications. Many
certificate (MIC) phones come from the factory with MICs, but MICs may expire or
locally significant certificate become lost or compromised. Some phones do not come with MICs.
(LSC) LSCs are certificates that are issued locally to the phones using the
CAPF server.
transport Layer Security (TLS) IETF standard (RFC 2246) protocol, based on Netscape Secure
protocol Socket Layer (SSL) protocol. TLS sessions are established using a
handshake protocol to provide privacy and data integrity.
The TLS record layer fragments and defragments, compresses and
decompresses, and performs encryption and decryption of application
data and other TLS information, including handshake messages.
Figure 19 shows the components in a Cisco Unified CME phone authentication environment.
Secondary
External CA Cisco Unified
server CME router
(optional) (optional)
CA CTL provider
Primary
Cisco Unified CME router
CTL
protocol
Cisco IOS CA Cisco IOS RA
CTL file
CTL client
Signed
configuration
Secure Telephony files
CAPF server TFTP store
SCCP server services module
Signaling Certificate
TFTP server
TLS TLS
146624
IP
phone downloads the new CTL file, it verifies the download with only one of the two original
public keys that was installed earlier. This mechanism prevents IP phones from accepting CTL
files from unknown sources.
c. The CTL file is published on the TFTP server. Because an external TFTP server is not supported
in secure mode, the configuration files are generated by the Cisco Unified CME system itself
and are digitally signed by the TFTP server’s credentials. The TFTP server credentials can be
the same as the Cisco Unified CME credentials. If desired, a separate certificate can be
generated for the TFTP function if the appropriate trustpoint is configured under the CTL-client
interface.
3. The telephony service module signs phone configuration files and each phone requests its file.
4. When an IP phone boots up, it requests the CTL file (CTLfile.tlv) from the TFTP server and
downloads its digitally signed configuration file, which has the filename format of
SEP<mac-address>.cnf.xml.sgn.
5. The phone then reads the CAPF configuration status from the configuration file. If a certificate
operation is needed, the phone initiates a TLS session with the CAPF server on TCP port 3804 and
begins the CAPF protocol dialogue. The certificate operation can be an upgrade, delete, or fetch
operation. If an upgrade operation is needed, the CAPF server makes a request on behalf of the
phone for a certificate from the CA. The CAPF server uses the CAPF protocol to obtain the
information it needs from the phone, such as the public key and phone ID. After the phone
successfully receives a certificate from the server, the phone stores it in its flash memory.
6. With the certificate in its flash, the phone initiates a TLS connection with the secure
Cisco Unified CME server on a well-known TCP port (2443), if the device security mode settings
in the .cnf.xml file are set to authenticated or encrypted. This TLS session is mutually authenticated
by both parties. The IP phone knows the Cisco Unified CME server’s certificate from the CTL file,
which it initially downloaded from the TFTP server. The phone’s LSC is a trusted party for the
Cisco Unified CME server, because the issuing CA certificate is present in the router.
Startup Messages
If the certificate server is part of your startup configuration, you may see the following messages during
the boot procedure:
% Failed to find Certificate Server's trustpoint at startup
% Failed to find Certificate Server's cert.
These messages are informational messages that show a temporary inability to configure the certificate
server because the startup configuration has not been fully parsed yet. The messages are useful for
debugging, if the startup configuration has been corrupted.
Other configuration files that are not generated by Cisco Unified CME, such as ringlist.xml,
distinctiveringlist.xml, audio files, and so forth, are often used for Cisco Unified CME features. Signed
versions of these configuration files are not automatically created. Whenever a new configuration file
that has not been generated by Cisco Unified CME is imported into Cisco Unified CME, use the
load-cfg-file command, which does all of the following:
• Hosts the unsigned version of the file on the TFTP server.
• Creates a signed version of the file.
• Hosts the signed version of the file on the TFTP server.
You can also use the load-cfg-file command instead of the tftp-server command when only the
unsigned version of a file needs to be hosted on the TFTP server.
H.323 gateway
VoIP
Cisco VG224 analog V IP
V phone gateway Nonsecure
H.323 secure phone D
DSP farm
U
Cisco Unity
V
Cisco Unified CME 1
Cisco Unity Express Cisco 800 series router
WAN
IP IP
170910
Local phone A Local phone B IP
Remote phone C
Secure Cisco Unified CME implements call control signaling using Transport Layer Security (TLS) or
IPsec (IP Security) for the secure channel, and uses SRTP for media encryption. Secure
Cisco Unified CME manages the SRTP keys to endpoints and to gateways.
The Media Encryption (SRTP) on Cisco Unified CME feature supports the following features:
• Secure voice calls using SRTP for SCCP endpoints
• Secure voice calls in a mixed shared line environment that allows both RTP and SRTP capable
endpoints; shared line media security depends on the endpoint configuration.
• Secure supplementary services using H.450 including:
– Call forward
– Call transfer
– Call hold and resume
– Call park and call pickup
– Nonsecure software conference
Note SRTP conference calls over H.323 may experience a 0 to 2 second noise interval when the call is joined
to the conference.
231361
Phone A Phone B Phone C
SRST-capable SRST-capable
3 A hears music
on hold
2 B initiates supplementary
1 A calls B services by calling
C to transfer the call
IP IP IP
231360
Phone A Phone B Phone C
The media path is optional. The default media path for Cisco Unified CME is hairpin. However,
whenever possible media flow around can be configured on Cisco Unified CME. When configuring
media flow through, which is the default, remember that chaining multiple XOR gateways in the media
path introduces more delay and thus reduces voice quality. Router resources and voice quality limit the
number of XOR gateways that can be chained. The requirement is platform dependent, and may vary
between signaling and media. The practical chaining level is three.
A transcoder is inserted when there is a codec mismatch and ECS and TCS negotiation fails. For
example, if Phone A and Phone B are SRTP capable, but Phone A uses the G.711 codec and Phone B
uses the G.729 codec, a transcoder is inserted if Phone B has one. However, the call is negotiated down
to RTP to fulfill the codec requirement, so the call is not secure.
Note Transcoding is enabled only if an H.323 call with a different codec from the remote phone tries to make
a call to the remote phone. If a local phone on the same Cisco Unified CME as the remote phone makes
a call to remote phone, the local phone is forced to change its codec to G.729 instead of using
transcoding.
Secure transcoding for point-to-point SRTP calls can only occur when both the SCCP phone which is to
be serviced by Cisco Unified CME transcoding and its peer in the call are SRTP-capable and have
successfully negotiated the SRTP keys. Secure transcoding for point-to-point SRTP calls cannot occur
when only one of the peers in the call is SRTP-capable.
If Cisco Unified CME transcoding is to be performed on a secure call, the Media Encryption (SRTP) on
Cisco Unified CME feature allows Cisco Unified CME to provide the DSP farm with the encryption
keys for the secure call as additional parameters, so that Cisco Unified CME transcoding can be
performed successfully. Without the encryption keys, the DSP farm would not be able to read the
encrypted voice data in order to transcode it.
Note The secure transcoding described here does not apply to IP-IP gateway transcoding.
Cisco Unified CME transcoding is different from IP-to-IP gateway transcoding because it is invoked for
an SCCP endpoint only, instead of for bridging VoIP call legs. Cisco Unified CME transcoding and
IP-to-IP gateway transcoding are mutually exclusive, that is, only one type of transcoding can be
invoked for a call. If no DSP farm capable of SRTP transcoding is available, Cisco Unified CME secure
transcoding is not performed and the call goes through using G.711.
Note Cisco Unity Express does not support secure signaling and media encryption. Secure
Cisco Unified CME interoperates with Cisco Unity Express, but calls between Cisco Unified CME and
Cisco Unity Express are not secure.
In a typical Cisco Unity Express deployment with Cisco Unified CME in a secure H.323 network,
Session Initiation Protocol (SIP) is used for signaling, and the media path is G.711 with RTP. For Call
Forward No Answer (CFNA) and Call Forward All (CFA), before the media path is established,
signaling messages are sent to negotiate an RTP media path. If codec negotiation fails, a transcoder is
inserted. The Media Encryption (SRTP) on Cisco Unified CME feature’s H.323 service provider
interface (SPI) supports fast start calls. In general, calls transferred or forwarded back to
Cisco Unified CME from Cisco Unity Express fall into existing call flows and are treated as regular SIP
and RTP calls.
The Media Encryption (SRTP) on Cisco Unified CME feature supports blind transfer back to
Cisco Unified CME only. When midcall media renegotiation is configured, the secure capability for the
endpoint is renegotiated regardless of which transfer mechanism, H.450.2 or Empty Capability Set
(ECS), was used.
Phone Authentication
• Configuring the Cisco IOS Certification Authority, page 402 (required)
• Verifying the Cisco IOS Certification Authority, page 406 (optional)
• Configuring the Registration Authority, page 406 (optional)
• Verifying the Registration Authority, page 409 (optional)
• Authenticating Certificates for Server Functions, page 409 (required)
• Verifying Certificates for Server Functions, page 412 (optional)
• Manually Importing MIC Root Certificate, page 412 (optional)
• Configuring Telephony-Service Security Parameters, page 414 (required)
• Verifying Telephony-Service Security Parameters, page 419 (optional)
• Configuring the CTL Client, page 419 (required)
• Verifying the CTL Client, page 424 (optional)
• Configuring the CTL Provider, page 424 (optional)
• Verifying the CTL Provider, page 426 (optional)
• Configuring the CAPF Server, page 426 (required)
• Verifying the CAPF Server, page 429 (optional)
• Entering the Authentication String on the Phone, page 430 (optional)
• Verifying the Authentication String on the Phone, page 431 (optional)
Media Encryption
• Configuring Secure Calls Between Cisco Unified CMEs Across an H.323 Trunk, page 431
(required)
• Configuring Cisco Unified CME SRTP Fallback for H.323 Dial Peers, page 433 (optional)
• Configuring Cisco Unity for Secure Cisco Unified CME Operation, page 435 (optional)
Note If you use a third-party CA, follow the provider’s instructions instead of performing these steps.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip http server
4. crypto pki server label
5. database level {minimal | names | complete}
6. database url root-url
7. lifetime certificate time
8. issuer-name CN=label
9. exit
10. crypto pki trustpoint label
11. enrollment url ca-url
12. exit
13. crypto pki server label
14. grant auto
15. no shutdown
16. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ip http server Enables the Cisco web-browser user interface on the local
Cisco Unified CME router.
Example:
Router(config)# ip http server
Step 4 crypto pki server label Defines a label for the certificate server and enters
certificate-server configuration mode.
Example: • label—Name for CA certificate server.
Router(config)# crypto pki server sanjose1
Example:
Router(config-cs-server)# exit
Step 10 crypto pki trustpoint label (Optional) Declares a trustpoint and enters ca-trustpoint
configuration mode.
Example: • label—Name for the trustpoint.
Router(config)# crypto pki trustpoint sanjose1
• A trustpoint for the CA is automatically generated by
the router when the CA is started. If you must use a
specific RSA key for the CA, you can create your own
trustpoint by using the same label used in the crypto
pki server command in Step 13. If the router sees a
configured trustpoint with the same label as that of the
“crypto pki server,” it uses this trustpoint and does not
automatically create a trustpoint.
Note Use this command and the enrollment url
command if this CA is local to the
Cisco Unified CME router. These commands are
not needed for a CA on an external router.
Step 11 enrollment url ca-url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • ca-url—URL of the router on which the root CA is
Router(config-ca-trustpoint)# enrollment url installed.
http://ca-server.company.com
Step 12 exit Exits ca-trustpoint configuration mode.
Example:
Router(config-ca-trustpoint)# exit
Step 13 crypto pki server label Enters certificate-server configuration mode.
• label—Name for CA certificate server.
Example:
Router(config)# crypto pki server sanjose1
Step 14 grant auto (Optional) Allows certificates to be issued automatically to
any requester.
Example: • Default and recommended method is manual
Router(config-cs-server)# grant auto enrollment.
Tip Use this command only when testing and building
simple networks. Use the no grant auto command
after configuration is complete to prevent
certificates from being automatically granted.
Example:
Router(config-cs-server)# end
To configure an RA, perform the following steps on the Cisco Unified CME router.
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki trustpoint label
4. enrollment url ca-url
5. revocation-check method1 [method2 [method3]]
6. serial-number [none]
7. rsakeypair key-label [key-size [encryption-key-size]]
8. exit
9. crypto pki server label
10. mode ra
11. lifetime certificate time
12. grant auto
13. no shutdown
14. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 crypto pki trustpoint label Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA. The
Router(config)# crypto pki trustpoint ra12 certificate-server label that you use here is also used in
the crypto pki server command in Step 9.
Note This name is also specified in the
cert-enroll-trustpoint command when you set up
the CA proxy as described in the “Configuring the
CAPF Server” section on page 426.
Step 4 enrollment url ca-url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • ca-url—URL of the router on which the root CA has
Router(config-ca-trustpoint)# enrollment url been installed.
http://ca-server.company.com
Example:
Router(config-ca-trustpoint)# exit
Step 9 crypto pki server label Defines a label for the certificate server and enters
certificate-server configuration mode.
Example: • label—Name for the trustpoint and RA. The
Router(config)# crypto pki server ra12 certificate-server label must have the same name as the
trustpoint that was created in Step 3.
Step 10 mode ra Places the PKI server into certificate-server mode for the
RA.
Example:
Router(config-cs-server)# mode ra
Example:
Router(config-cs-server)# end
To obtain a certificate for each of these functions, perform the following steps for each server function.
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki trustpoint trustpoint-label
4. enrollment url url
5. revocation-check method1 [method2 [method3]]
6. rsakeypair key-label [key-size [encryption-key-size]]
7. exit
8. crypto pki authenticate trustpoint-label
9. crypto pki enroll trustpoint-label
10. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 crypto pki trustpoint trustpoint-label Declares the trustpoint that the Cisco Unified CME
certificate server should use and enters ca-trustpoint
configuration mode.
Example:
Router(config)# crypto pki trustpoint capf • trustpoint-label—Label for the trustpoint.
Step 4 enrollment url url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • url—URL of the router on which the root CA is
Router(config-ca-trustpoint)# enrollment url installed.
http://ca-server.company.com
Example:
Router(config-ca-trustpoint)# exit
Step 8 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate capf
Note This command is optional if the CA certificate is
already loaded into the configuration.
Step 9 crypto pki enroll trustpoint-label Enrolls with the CA and obtains the certificate for this
trustpoint.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki enroll capf
Step 10 exit Returns to privileged EXEC mode.
Example:
Router(config)# exit
Prerequisites
One of the following situations must be true before you perform this task:
• You choose to use MIC as the method for phone authentication during CAPF certificate operation
• You plan to establish the TLS session for SCCP signaling using the phone’s MIC instead of an LSC
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki trustpoint name
4. revocation-check method1
5. enrollment terminal
6. exit
7. crypto pki authenticate name
8. Open the MIC root file and copy the certificate.
9. When prompted, paste the certificate, press Enter, and type quit.
10. Enter y to accept the certificate.
11. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 crypto pki trustpoint name Declares the CA that your router should use and enters
CA-trustpoint configuration mode.
Example: • name—CA trustpoint name.
Router(config)# crypto pki trustpoint sanjose1
Step 4 revocation-check method1 Checks the revocation status of a certificate.
• method1—The method used by the router to check the
Example: revocation status of the certificate. For this task, the
Router(ca-trustpoint)# revocation-check none only available method is none. The keyword none is
required for this task and means that a revocation check
is not performed and the certificate is always accepted.
Step 5 enrollment terminal Specifies manual (copy-and-paste) certificate enrollment.
Example:
Router(ca-trustpoint)# enrollment terminal
Step 6 exit Exits CA-trustpoint configuration mode.
Example:
Router(ca-trustpoint)# exit
Step 7 crypto pki authenticate name Authenticates the CA (by getting the certificate from the
CA).
Example: • name—Name of the CA.
Router(config)# crypto pki authenticate
sanjose1
Step 8 Open the MIC root file and copy the certificate. The MIC root file is a file with name a*.0, located in the
directory C:\Program Files\Cisco\Certificates
Copy to a buffer or temporary location all of the contents
that appear between “-----BEGIN CERTIFICATE-----” and
“-----END CERTIFICATE-----”.
Step 9 When prompted, paste the certificate, press Enter, and Paste the text from the a*.0 file, press Enter after pasting the
type quit. certificate, and type quit on a line by itself.
Example:
Router(config)# exit
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. secure-signaling trustpoint label
5. tftp-server-credentials trustpoint label
6. device-security-mode {authenticated | none | encrypted}
7. cnf-file perphone
8. load-cfg-file file-url alias file-alias [sign] [create]
9. server-security-mode {secure | non-secure}
10. exit
11. ephone phone-tag
12. device-security-mode {authenticated | none | encrypted}
13. codec {g711ulaw | g729r8 [dspfarm-assist]}
14. capf-auth-str digit-string
15. cert-oper {delete | fetch | upgrade} auth-mode {auth-string | LSC | MIC | null-string}
16. reset
17. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 secure-signaling trustpoint label Specifies the name of the PKI trustpoint that has the valid
certificate to be used for TLS handshakes with IP phones on
TCP port 2443.
Example:
Router(config-telephony)# secure-signaling • label—Name of a configured PKI trustpoint with a
trustpoint cme-sccp valid certificate.
Step 5 tftp-server-credentials trustpoint label Specifies the name of the PKI trustpoint to be used to sign
the phone configuration files. This can be the CAPF-server
trustpoint that was used in the previous step or any
Example:
Router(config-telephony)#
trustpoint with a valid certificate.
tftp-server-credentials trustpoint cme-tftp • label—Name of a configured PKI trustpoint with a
valid certificate.
Step 6 device-security-mode {authenticated | none | Enables security mode for all security-capable phones in the
encrypted} system.
• authenticated—SCCP signaling between a device and
Example: Cisco Unified CME takes place through the secure TLS
Router(config-telephony)# device-security-mode connection on TCP port 2443.
authenticated
• none—SCCP signaling is not secure. This is the
default.
• encrypted—SCCP signaling between a device and
Cisco Unified CME takes place through the secure TLS
connection on TCP port 2443, and the media uses
Secure Real-Time Transport Protocol (SRTP). Use the
encrypted keyword to enable Secure
Cisco Unified CME functionality.
Note You can override the setting you make in this
command for individual ephones by using the
device-security-mode command in ephone
configuration mode.
Example:
Router(config)# exit
Step 11 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Identifier of the ephone to be configured.
Example:
Router(config)# ephone 24
Example:
Router(config-ephone)# reset
Step 17 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
telephony-service
secure-signaling trustpoint cme-sccp
server-security-mode secure
device-security-mode authenticated
tftp-server-credentials trustpoint cme-tftp
.
.
.
SUMMARY STEPS
1. enable
2. configure terminal
3. ctl-client
4. sast1 trustpoint trustpoint-label
5. sast2 trustpoint trustpoint-label
6. server {capf | cme | cme-tftp | tftp} ip-address trustpoint trustpoint-label
7. server cme ip-address username string password 0 string
8. regenerate
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ctl-client Enters CTL-client configuration mode.
Example:
Router(config)# ctl-client
Step 4 sast1 trustpoint label Configures credentials for the primary SAST.
• label—SAST1 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast1 trustpoint
sast1tp
from each other. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.
Example:
Router(config-ctl-client)# end
What to do Next
When you have more than one Cisco Unified CME router in your network, you must configure a CTL
provider on each Cisco Unified CME router that is not running the CTL client. See the “Configuring the
CTL Provider” section on page 424.
Configuring the CTL Client on a Router Other Than a Cisco Unified CME Router
To configure a CTL client on an external router that is not a Cisco Unified CME router, perform the
following steps.
SUMMARY STEPS
1. enable
2. configure terminal
3. ctl-client
4. sast1 trustpoint trustpoint-label
5. sast2 trustpoint trustpoint-label
6. server cme ip-address username name-string password {0 | 1} password-string
7. regenerate
8. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# ctl-client
Step 4 sast1 trustpoint label Configures credentials for the primary SAST.
• label—SAST1 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast1 trustpoint
sast1tp
from each other, but either of them may use the
same certificate as the Cisco Unified CME router to
conserve memory. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.
Step 5 sast2 trustpoint label Configures credentials for the secondary SAST.
• label—SAST2 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast2 trustpoint
from each other, but either of them may use the
same certificate as the Cisco Unified CME router to
conserve memory. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.
Step 6 server cme ip-address username name-string (Optional) Provides information about another
password {0 | 1} password-string Cisco Unified CME router (primary or secondary) in the
network, if one exists.
Example: • ip-address—IP address of the other
Router(config-ctl-client)# server cme 10.2.2.2 Cisco Unified CME router.
username user3 password 0 38h2KL
• username name-string—Username that is configured
on the CTL provider.
• password—Encryption status of the password string.
– 0—Not encrypted.
– 1—Encrypted using Message Digest 5 (MD5).
Note This option refers to the way that you want the
password to appear in show command output and
not to the way that you enter the password in this
command.
Example:
Router(config-ctl-client)# end
What to do Next
You must configure a CTL provider on each Cisco Unified CME router. See the “Configuring the CTL
Provider” section on page 424.
SUMMARY STEPS
1. enable
2. configure terminal
3. credentials
4. ip source-address ip-address port port-number
5. trustpoint trustpoint-label
6. ctl-service admin username secret {0 | 1} password-string
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 credentials Enters credentials-interface mode to configure a CTL
provider.
Example:
Router(config)# credentials
Step 4 ip source-address [ip-address [port Identifies the local router on which this CTL provider is
[port-number]]] being configured.
• ip-address—Router IP address, typically one of the
Example: addresses of the Ethernet port of the router.
Router(config-credentials)# ip source-address
172.19.245.1 port 2444 • port port-number—TCP port for credentials service
communication. Default is 2444. You should use 2444.
Step 5 trustpoint trustpoint-label Configures the trustpoint to be used for TLS sessions with
the CTL client.
Example: • trustpoint-label—CTL provider trustpoint label.
Router(config-credentials)# trustpoint ctlpv
Step 6 ctl-service admin username secret {0 | 1} Specifies a username and password to authenticate the CTL
password-string client when it connects to retrieve the credentials during the
CTL protocol. You must use this command before you
Example: enable the CTL provider.
Router(config-credentials)# ctl-service admin • username—Name that will be used to authenticate the
user4 secret 0 c89L8o
client.
• secret—Character string for login authentication and
whether the string should be encrypted when it is stored
in the running configuration.
– 0—Not encrypted.
– 1—Encrypted using Message Digest 5 (MD5).
• password-string—Character string for login
authentication.
Step 7 end Returns to privileged EXEC mode.
Example:
Router(config-credentials)# end
Tip When you use the CAPF server to install phone certificates, arrange to do so during a scheduled period
of maintenance. Generating many certificates at the same time may cause call-processing interruptions.
SUMMARY STEPS
1. enable
2. configure terminal
3. capf-server
4. trustpoint-label label
5. cert-enroll-trustpoint label password {0 | 1} password-string
6. source-addr ip-address
7. port tcp-port
8. auth-mode {auth-string | LSC | MIC | none | null-string}
9. auth-string {delete | generate} {all | ephone-tag} [auth-string]
10. phone-key-size {512 | 1024 | 2048}
11. keygen-retry number
12. keygen-timeout minutes
13. cert-oper {delete all | fetch all | upgrade all}
14. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 capf-server Enters CAPF-server configuration mode.
Example:
Router(config)# capf-server
Step 4 trustpoint-label label Specifies the label of the trustpoint whose certificate is to be
used for TLS connection between the CAPF server and the
phone.
Example:
Router(config-capf-server)# trustpoint-label • label—Trustpoint name.
tp1
Step 5 cert-enroll-trustpoint trustpoint-label Enrolls the CAPF with the CA (or RA if the CA is not local
password {0 | 1} password-string to the Cisco Unified CME router).
• trustpoint-label—PKI trustpoint label for the CA or
Example: RA.
Router(config-capf-server)#
cert-enroll-trustpoint ra1 password 0 x8oWiet • password—Encryption status of the password string.
• password-string—Password to use for certificate
enrollment. This password is the revocation password
that is sent along with the certificate request to the CA.
Step 6 source-addr ip-address Defines the IP address of the CAPF server on the
Cisco Unified CME router.
Example: • ip-address—IP address of the CAPF server.
Router(config-capf-server)# source addr
10.10.10.1
Step 7 port tcp-port (Optional) Defines the TCP port number on which the
CAPF server listens for socket connections from the
phones.
Example:
Router(config-capf-server)# port 3804 • tcp-port—TCP port number. Range is 2000 to 9999.
Default is 3804.
Example:
Router(config-capf-server)# end
What to Do Next
If you select the authentication-string method of authentication in the auth-mode command, you must
also enter an authentication string on each phone that is receiving an updated LSC. For instructions on
this task, see the “Entering the Authentication String on the Phone” section on page 430.
Note You can list authentication strings for phones by using the show capf-server auth-string command.
Prerequisites
• The CAPF certificate exists in the CTL file.
• A signed image exists on the phone; see the Cisco Unified IP phone administration documentation
that supports your phone model.
• The device has registered.
• The device security mode is nonsecure.
DETAILED STEPS
Step 5 When prompted for the authentication string, enter the string provided by the system administrator and
press the Submit soft key.
The phone installs, updates, deletes, or fetches the certificate, depending on the CAPF configuration.
You can monitor the progress of the certificate operation by viewing the messages that display on the
phone. After you press Submit, the message “Pending” displays under the LSC option. The phone
generates the public and private key pair and displays the information on the phone. When the phone
successfully completes the process, the phone displays a successful message. If the phone displays a
failure message, you entered the wrong authentication string or did not enable the phone for upgrade.
You can stop the process by choosing the Stop option at any time.
Configuring Secure Calls Between Cisco Unified CMEs Across an H.323 Trunk
To configure the network for secure calls between Cisco Unified CME systems across an H.323 trunk,
perform the following steps on the Cisco Unified CME router.
Prerequisites
To make secure H.323 calls, telephony-service security parameters must be configured. See the
“Configuring Telephony-Service Security Parameters” section on page 414.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. supplementary-service media-renegotiate
5. srtp fallback
6. h323
7. emptycapability
8. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode.
• The voip keyword specifies VoIP encapsulation.
Example:
Router(config)# voice service voip
Step 4 supplementary-service media-renegotiate Enables midcall renegotiation of SRTP cryptographic keys.
Example:
Router(conf-voi-serv)# supplementary-service
media-renegotiate
Step 5 srtp fallback Enables security policies.
• The srtp command enables secure calls using SRTP for
Example: media encryption and authentication and disables
Router(conf-voi-serv)# srtp fallback fallback.
• The fallback keyword enables call fallback to
nonsecure (RTP) mode, allowing the user to make calls
that are not secure.
• SRTP-to-RTP fallback must be configured for
supplementary services such as ringback tone and
MOH to function. Without SRTP-to-RTP fallback
configured, MOH causes secure calls to be dropped.
Note This security policy applies to all calls going
through the gateway and is not configurable on a
per-call basis.
If fallback is not configured it will drop all calls that are not
secure so only secure phones can call you.
This step configures fallback globally. To configure
fallback for individual dial peers, see the “Configuring
Cisco Unified CME SRTP Fallback for H.323 Dial Peers”
section on page 433. Skip this step if you are going to
configure fallback on individual dial peers.
Step 6 h323 Enters H.323 voice-service configuration mode.
Example:
Router(conf-voi-serv)# h323
Example:
Router(conf-serv-h323)# exit
Configuring Cisco Unified CME SRTP Fallback for H.323 Dial Peers
To configure SRTP fallback for an individual dial peer, perform the following steps on the
Cisco Unified CME router.
Note SRTP-to-RTP fallback must be configured for supplementary services such as ringback tone and MOH
to function. Without SRTP-to-RTP fallback configured, MOH causes secure calls to be dropped.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice class codec tag
4. codec preference value codec-type
5. exit
6. dial-peer voice tag voip
7. srtp fallback
8. voice-class codec tag
9. exit
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-voice-class)# exit
Step 6 dial-peer voice tag voip Enters dial peer voice configuration mode.
Example:
Router(config)# dial-peer voice 101 voip
Step 7 srtp fallback Enables secure calls that use SRTP for media encryption
and authentication and specifies fallback capability. Using
the no srtp command disables security and causes the dial
Example:
Router(config-dial-peer)# srtp fallback
peer to fall back to RTP mode.
• The srtp command enables secure calls.
• The fallback keyword enables fallback to nonsecure
mode (RTP) on an individual dial peer. The no form of
this command disables fallback and disables SRTP.
Note This dial-peer configuration command takes
precedence over the globally configured srtp
command enabled in voice service voip
configuration mode shown in the “Configuring
Secure Calls Between Cisco Unified CMEs Across
an H.323 Trunk” section on page 431.
Step 8 voice-class codec tag Assigns a previously configured codec selection preference
list (codec voice class) to a Voice over IP (VoIP) dial peer.
Example: • The tag argument in this step is the same as the tag in
Router(config-dial-peer)# voice-class codec 1 Step 3.
Step 9 exit Exits dial-peer voice configuration mode.
Example:
Router(config-dial-peer)# exit
Step 1 If Cisco Unity Telephony Integration Manager (UTIM) is not already open, on the Cisco Unity server,
on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM
window appears.
Step 2 In the left pane, double-click Cisco Unity Server. The existing integrations appear.
Step 3 Click the Cisco Unified Communications Manager integration.
Step 4 In the right pane, click the cluster for the integration.
Step 5 Click the Servers tab.
Step 6 In the Cisco Unified Communications Manager Cluster Security Mode field, click the applicable setting.
Step 7 If you clicked the Non-secure setting, click Save and skip the remaining steps in this procedure.
If you clicked the Authenticated or the Encrypted settings, the Security tab and the Add TFTP Server
dialog box appear. In the Add TFTP Server dialog box, in the IP Address or Host Name field, enter the
IP address (or DNS name) of the primary TFTP server for the Cisco Unified Communications Manager
cluster, and click OK.
Step 8 If there are more TFTP servers that Cisco Unity will use to download the Cisco Unified Communications
Manager certificates, click Add. The Add TFTP Server dialog box appears.
Step 9 In the IP Address or Host Name field, enter the IP address (or DNS name) of the secondary TFTP server
for the Cisco Unified Communications Manager cluster, and click OK.
Step 10 Click Save.
Cisco Unity creates the voice messaging port device certificates, exports the Cisco Unity server root
certificate, and displays the Export Cisco Unity Root Certificate dialog box.
Step 11 Note the file name of the exported Cisco Unity server root certificate and click OK.
Step 12 On the Cisco Unity server, navigate to the CommServer\SkinnyCerts directory.
Step 13 Locate the Cisco Unity server root certificate file that you exported in Step 11.
Step 14 Right-click the file and click Rename.
Step 15 Change the file extension from .0 to .pem. For example, change the filename “12345.0” to “12345.pem”
for the exported Cisco Unity server root certificate file.
Step 16 Copy this file to a PC from which you can access the Cisco Unified CME router.
SUMMARY STEPS
1. enable
2. configure terminal
3. crypto pki trustpoint name
4. revocation-check none
5. enrollment terminal
6. exit
7. crypto pki authenticate trustpoint-label
8. Open the root certificate file that you copied from the Cisco Unity Server in Step 16.
9. You will be prompted to enter the CA certificate. Cut and paste the entire contents of the base 64
encoded certificate between “BEGIN CERTIFICATE” and “END CERTIFICATE” at the command
line. Press Enter, and type “quit.” The router prompts you to accept the certificate. Enter “yes” to
accept the certificate.
DETAILED STEPS
Example:
Router# configure terminal
Step 3 crypto pki trustpoint name Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA.
Router(config)# crypto pki trustpoint PEM
Step 4 revocation-check none (Optional) Checks the revocation status of a certificate and
specifies one or more methods to check the status. If a
second and third method are specified, each method is used
Example:
Router(ca-trustpoint)# revocation-check none
only if the previous method returns an error, such as a server
being down.
• none—Certificate checking is not required.
Step 5 enrollment terminal Specifies manual cut-and-paste certificate enrollment.
Example:
Router(ca-trustpoint)# enrollment terminal
Example:
Router(ca-trustpoint)# exit
Step 7 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate pem
Note The trustpoint-label must be the same as the name
in step 3.
Step 8 You will be prompted to enter the CA certificate. Cut Completes the copying of the Cisco Unity root certificate to
and paste the entire contents of the base 64 encoded the Cisco Unified CME router.
certificate between “BEGIN CERTIFICATE” and
“END CERTIFICATE” at the command line. Press
Enter, and type “quit.” The router prompts you to
accept the certificate. Enter “yes” to accept the
certificate.
Step 1 Choose the Cisco voice-mail port that you want to update.
Step 2 In the Device Security Mode field, choose Encrypted from the drop-down list box.
Step 3 Click Update.
In the following example, the secure value of the stype field shows that the connections are secure.
Router# show sccp connections
Phone Authentication
• Cisco IOS CA Server: Example, page 438
• Enabling a Registration Authority: Example, page 438
• Manually Importing MIC Root Certificate on the Cisco Unified CME Router: Example, page 439
• Obtaining a Certificate for Cisco Unified CME Server Functions: Example, page 442
• CTL Client Running on Cisco Unified CME Router: Example, page 442
• CTL Client Running on Another Router: Example, page 442
• Telephony-Service Security Parameters: Example, page 442
• CAPF Server: Example, page 443
Media Encryption
• Secure Cisco Voice Gateway with Media Encryption (SRTP) on Cisco Unified CME: Example,
page 445
• Secure Cisco Unified CME: Example, page 451
The following example sets up a trustpoint named sast2 that periodically generates a CRL instead of
having it generated manually. Third-party CAs may require this functionality.
Router(config)# crypto pki trustpoint sast2
Router(config-ca-trustpoint)# enrollment url http://NTP-ab11:80
Router(config-ca-trustpoint)# serial-number
Router(config-ca-trustpoint)# revocation-check crl
Router(config-ca-trustpoint)# rsakeypair sast2
Manually Importing MIC Root Certificate on the Cisco Unified CME Router:
Example
The following example shows three certificates imported to the router (7970, 7960, PEM).
Router(config)# crypto pki trustpoint 7970
Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate 7970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quit
Certificate has the following attributes:
Fingerprint MD5: 4B9636DF 0F3BA6B7 5F54BE72 24762DBC
Fingerprint SHA1: A9917775 F86BB37A 5C130ED2 3E528BB8 286E8C2D
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported
Use the show crypto pki trustpoint status command to show that enrollment has succeeded and that
five CA certificates were granted. The five certificates include the three certificates just entered and the
CA server certificate and the router certificate.
Router# show crypto pki trustpoint status
Trustpoint 7970:
Issuing CA certificate configured:
Subject Name:
cn=CAP-RTP-002,o=Cisco Systems
Fingerprint MD5: F7E150EA 5E6E3AC5 615FC696 66415C9F
Fingerprint SHA1: 1BE2B503 DC72EE28 0C0F6B18 798236D8 D3B18BE6
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint 7960:
Issuing CA certificate configured:
Subject Name:
cn=CAPF-508A3754,o=Cisco Systems Inc,c=US
Fingerprint MD5: 6BAE18C2 0BCE391E DAE2FE4C 5810F576
Fingerprint SHA1: B7735A2E 3A5C274F C311D7F1 3BE89942 355102DE
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint PEM:
Issuing CA certificate configured:
Subject Name:
cn=CAP-RTP-001,o=Cisco Systems
Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint srstcaserver:
Issuing CA certificate configured:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint srstca:
Issuing CA certificate configured:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
Router General Purpose certificate configured:
Subject Name:
serialNumber=F3246544+hostname=c2611XM-sSRST.cisco.com
Fingerprint: 35471295 1C907EC1 45B347BC 7A9C4B86
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... Yes
ephone 24
device-security-mode authenticated
capf-auth-str 2734
cert-oper upgrade auth-mode auth-string
!
ephone 3
device-security-mode encrypted
capf-auth-str 5425
cert-oper upgrade auth-mode null-string
mac-address 000D.299D.50DF
type 7970
button 1:3
!
!
ephone 4
device-security-mode encrypted
capf-auth-str 7176
cert-oper upgrade auth-mode null-string
mac-address 000E.D7B1.0DAC
type 7960
button 1:4
!
!
ephone 5
device-security-mode encrypted
mac-address 000F.9048.5077
type 7960
button 1:5
!
!
ephone 6
device-security-mode encrypted
mac-address 0013.C352.E7F1
type 7941GE
button 1:6
!
Building configuration...
resource policy
!
clock timezone PST -8
clock summer-time PDT recurring
no ip domain lookup
!
!
!
! The following lines show STCAPP security enabled at the system level:
stcapp ccm-group 1
stcapp security trustpoint analog
stcapp security mode encrypted
stcapp
!
voice-card 0
dsp services dspfarm
!
crypto pki trustpoint analog
enrollment url http://10.4.177.51:80
serial-number
revocation-check none
certificate ca 01
30820201 3082016A A0030201 02020101 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 756E6974 69746573 74301E17 0D303630 35303132
33303130 335A170D 30393034 33303233 30313033 5A301431 12301006 03550403
1309756E 69746974 65737430 819F300D 06092A86 4886F70D 01010105 0003818D
00308189 02818100 C2D07857 B8DF7F55 3C2365B3 2E1524CF EE898D1F D7A04075
D36F0229 392803DF B45246B4 A447506F A3FCDD00 9FC93CD7 5B5573E0 7BFD25E1
AB2F24E2 740D5765 7F628B6E 0FD39BEE 940D80FF 3B9F9F17 7ACA8F82 1A9E3179
458781E8 87C95E1B 17E6A61C 7D138AC1 D8E30F3C 88BFAFEE A94D5F8C E433DF71
F076E96C 9BB5327F 02030100 01A36330 61300F06 03551D13 0101FF04 05300301
01FF300E 0603551D 0F0101FF 04040302 0186301F 0603551D 23041830 168014B5
418287D0 61FE277C 9A1862B3 673BF7F7 0E47DD30 1D060355 1D0E0416 0414B541
8287D061 FE277C9A 1862B367 3BF7F70E 47DD300D 06092A86 4886F70D 01010405
00038181 002BB76E 22A59D73 6DBB62BA BAC3D5B4 2F739A26 D5FFF911 EDEB9BDC
7B29FECC E0B68E0F 22A3C0D0 8BA64592 30C6B628 5EFA3905 1B13BFE7 7CEB1456
55214435 07F752A6 73D5646A 4BB7B3C2 61E2C185 3A638FCA AE5AC6A1 3DB3590B
C3C6C924 D1E1E365 FE041B07 F3E2AF24 3701B664 A7879229 AFDF163A 00AA12AA
85866101 53
quit
crypto pki certificate chain analog
certificate 0A
308201BF 30820128 A0030201 0202010A 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 756E6974 69746573 74301E17 0D303630 35333032
31313630 345A170D 30373035 33303231 31363034 5A302A31 28301206 03550405
130B4648 4B303930 37463050 47301206 092A8648 86F70D01 09021605 616B6173
68305C30 0D06092A 864886F7 0D010101 0500034B 00304802 4100A6AD 0A376A6C
9EB668CC D0DF2A17 180E6CA2 FA5F243B 861EAA29 BE5FC488 A22AD4E8 5DFC22AC
13B43337 2F9FBA64 14E838EA 888E79DE 93AB63E4 4B4E2ECD 256D0203 010001A3
4F304D30 0B060355 1D0F0404 030205A0 301F0603 551D2304 18301680 14B54182
87D061FE 277C9A18 62B3673B F7F70E47 DD301D06 03551D0E 04160414 34D2D41C
274AB6E3 71A3A32C EC19D533 D3C0A020 300D0609 2A864886 F70D0101 04050003
818100A2 3947B1D0 FC5E9B79 0C1A28E7 BCB34C6C BB68C5F6 356F3F61 7525053E
0AED7325 9F286888 887810A6 B62FBAF3 BDC81542 C9828BBF 6A9FE936 AD3ED33B
D4F5AD22 E703C8E0 C3DDEAC8 2097A209 542551F7 6340A2A4 55A25A99 6A87367F
A0CBD9B6 E38D5E40 6479EB71 EFA644B3 93222D6F 235039AE BB9AA7B7 B1D07B3C FC6339
quit
certificate ca 01
30820201 3082016A A0030201 02020101 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 756E6974 69746573 74301E17 0D303630 35303132
33303130 335A170D 30393034 33303233 30313033 5A301431 12301006 03550403
1309756E 69746974 65737430 819F300D 06092A86 4886F70D 01010105 0003818D
00308189 02818100 C2D07857 B8DF7F55 3C2365B3 2E1524CF EE898D1F D7A04075
D36F0229 392803DF B45246B4 A447506F A3FCDD00 9FC93CD7 5B5573E0 7BFD25E1
!
voice-port 2/13
!
voice-port 2/14
!
voice-port 2/15
!
voice-port 2/16
!
voice-port 2/17
!
voice-port 2/18
!
voice-port 2/19
!
voice-port 2/20
!
voice-port 2/21
!
voice-port 2/22
!
voice-port 2/23
description choctaw
!
!
!
sccp local FastEthernet0/0
sccp ccm 10.4.177.51 identifier 1 version 4.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
!
dial-peer voice 5001 pots
service stcapp
port 2/0
!
dial-peer voice 5002 pots
service stcapp
! The following line shows the security mode configured on the dial peer.
security mode authenticated
port 2/1
!
dial-peer voice 5003 pots
service stcapp
security mode none
port 2/2
!
dial-peer voice 2000 voip
destination-pattern 7...
session target ipv4:10.4.177.100
incoming called-number 7000
codec g711ulaw
!
dial-peer voice 1 pots
!
dial-peer voice 5004 pots
service stcapp
shutdown
port 2/3
!
dial-peer voice 5005 pots
shutdown
destination-pattern 3001
port 2/4
!
dial-peer voice 5006 pots
service stcapp
shutdown
port 2/5
!
dial-peer voice 5007 pots
service stcapp
shutdown
port 2/6
!
dial-peer voice 5008 pots
service stcapp
shutdown
port 2/7
!
dial-peer voice 5009 pots
service stcapp
shutdown
port 2/8
!
dial-peer voice 5010 pots
service stcapp
shutdown
port 2/9
!
dial-peer voice 5011 pots
service stcapp
shutdown
port 2/10
!
dial-peer voice 5012 pots
service stcapp
shutdown
port 2/11
!
dial-peer voice 5013 pots
service stcapp
shutdown
port 2/12
!
dial-peer voice 5014 pots
service stcapp
shutdown
port 2/13
!
dial-peer voice 5015 pots
service stcapp
shutdown
port 2/14
!
dial-peer voice 5016 pots
service stcapp
shutdown
port 2/15
!
dial-peer voice 5017 pots
service stcapp
shutdown
port 2/16
!
Building configuration...
!
capf-server
port 3084
auth-mode null-string
cert-enroll-trustpoint iosra password 1 mypassword
trustpoint-label mytrustpoint1
source-addr 10.13.32.11
phone-key-size 512
!
voice call debug full-guid
!
voice service voip
srtp fallback
allow-connections h323 to h323
no supplementary-service h450.2
no supplementary-service h450.3
no supplementary-service h450.7
supplementary-service media-renegotiate
h323
emptycapability
ras rrq ttl 4000
!
!
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
voice class codec 3
codec preference 1 g729r8
codec preference 8 g711alaw
codec preference 9 g711ulaw
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g728
codec preference 3 g723ar63
codec preference 4 g711ulaw
!
!
voice iec syslog
voice statistics type iec
voice statistics time-range since-reset
!
!
!
crypto pki server myra
database level complete
grant auto
lifetime certificate 1800
!
crypto pki trustpoint myra
enrollment url http://10.13.32.11:80
revocation-check none
rsakeypair iosra
!
crypto pki trustpoint mytrustpoint1
enrollment url http://10.13.32.11:80
revocation-check none
rsakeypair mytrustpoint1
!
crypto pki trustpoint sast2
enrollment url http://10.13.32.11:80
revocation-check none
rsakeypair sast2
!
!
crypto pki certificate chain myra
certificate ca 01
308201F9 30820162 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343031
375A170D 30393037 30363035 34303137 5A301031 0E300C06 03550403 1305696F
73726130 819F300D 06092A86 4886F70D 01010105 0003818D 00308189 02818100
D8CE29F9 C9FDB1DD 0E1517E3 6CB4AAF7 52B83DE2 1C017ACA DFC4AF42 F9D10D08
E74BF95B 29378902 B49E32C4 85907384 84CAE4B2 7759BB84 8AB1F578 580793C4
B11A2DBE B2ED02CC DA0C3824 A5FCC377 18CE87EA C0C297BA BE54530F E62247D8
1483CD14 9FD89EFE 05DFBB37 E03FD3F8 B2B1C0B8 A1931BCC B1174A9E 6566F8F5
02030100 01A36330 61300F06 03551D13 0101FF04 05300301 01FF300E 0603551D
0F0101FF 04040302 0186301F 0603551D 23041830 168014B7 16F6FD67 29666C90
D0C62515 E14265A9 EB256230 1D060355 1D0E0416 0414B716 F6FD6729 666C90D0
C62515E1 4265A9EB 2562300D 06092A86 4886F70D 01010405 00038181 002B7F41
64535A66 D20D888E 661B9584 5E3A28DF 4E5A95B9 97E57CAE B07A7C38 7F3B60EE
75C7E5DE 6DF19B06 5F755FB5 190BABFC EF272CEF 865FE01B 1CE80F98 F320A569
CAFFA5D9 3DB3E7D8 8A86C66C F227FF81 6C4449F2 AF8015D9 8129C909 81AFDC01
180B61E8 85E19873 96DB3AE3 E6B70726 9BF93521 CA2FA906 99194ECA 8F
quit
crypto pki certificate chain mytrustpoint1
certificate 02
308201AB 30820114 A0030201 02020102 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343233
385A170D 30393037 30363035 34303137 5A301A31 18301606 092A8648 86F70D01
09021609 32383531 2D434D45 32305C30 0D06092A 864886F7 0D010101 0500034B
00304802 4100B3ED A902646C 3851B7F6 CF94887F 0EC437E3 3B6FEDB2 2B4B45A6
3611C243 5A0759EA 1E8D96D1 60ABE028 ED6A3F2A E95DCE45 BE0921AF 82E53E57
17CC12F0 C1270203 010001A3 4F304D30 0B060355 1D0F0404 030205A0 301F0603
551D2304 18301680 14B716F6 FD672966 6C90D0C6 2515E142 65A9EB25 62301D06
03551D0E 04160414 4EE1943C EA817A9E 7010D5B8 0467E9B0 6BA76746 300D0609
2A864886 F70D0101 04050003 81810003 564A6DA1 868B2669 7C096F9A 41173CFC
E49246EE C645E30B A0753E3B E1A265D1 6EA5A829 F10CD0E8 3F2E3AD4 39D8DFE8
83525F2B D19F5E15 F27D6262 62852D1F 43629B68 86D91B5F 7B2E2C25 3BD2CCC3
00EF4028 714339B2 6A7E0B2F 131D2D9E 0BE08853 5CCAE47C 4F74953C 19305A20
B2C97808 D6E01351 48366421 A1D407
quit
certificate ca 01
308201F9 30820162 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343031
375A170D 30393037 30363035 34303137 5A301031 0E300C06 03550403 1305696F
73726130 819F300D 06092A86 4886F70D 01010105 0003818D 00308189 02818100
D8CE29F9 C9FDB1DD 0E1517E3 6CB4AAF7 52B83DE2 1C017ACA DFC4AF42 F9D10D08
E74BF95B 29378902 B49E32C4 85907384 84CAE4B2 7759BB84 8AB1F578 580793C4
B11A2DBE B2ED02CC DA0C3824 A5FCC377 18CE87EA C0C297BA BE54530F E62247D8
1483CD14 9FD89EFE 05DFBB37 E03FD3F8 B2B1C0B8 A1931BCC B1174A9E 6566F8F5
02030100 01A36330 61300F06 03551D13 0101FF04 05300301 01FF300E 0603551D
0F0101FF 04040302 0186301F 0603551D 23041830 168014B7 16F6FD67 29666C90
D0C62515 E14265A9 EB256230 1D060355 1D0E0416 0414B716 F6FD6729 666C90D0
C62515E1 4265A9EB 2562300D 06092A86 4886F70D 01010405 00038181 002B7F41
64535A66 D20D888E 661B9584 5E3A28DF 4E5A95B9 97E57CAE B07A7C38 7F3B60EE
75C7E5DE 6DF19B06 5F755FB5 190BABFC EF272CEF 865FE01B 1CE80F98 F320A569
CAFFA5D9 3DB3E7D8 8A86C66C F227FF81 6C4449F2 AF8015D9 8129C909 81AFDC01
180B61E8 85E19873 96DB3AE3 E6B70726 9BF93521 CA2FA906 99194ECA 8F
quit
crypto pki certificate chain sast2
certificate 03
308201AB 30820114 A0030201 02020103 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343331
375A170D 30393037 30363035 34303137 5A301A31 18301606 092A8648 86F70D01
09021609 32383531 2D434D45 32305C30 0D06092A 864886F7 0D010101 0500034B
00304802 4100C703 840B11A7 81FCE5AE A14FE593 5114D3C2 5473F488 B8FB4CC5
41EAFA3A D99381D8 21AE6AA9 BA83A84E 9DF3E8C6 54978787 5EF6CC35 C334D55E
interface Serial1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
interface Serial1/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
ip route 0.0.0.0 0.0.0.0 10.13.32.1
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
!
!
!
!
tftp-server flash:music-on-hold.au
tftp-server flash:TERM70.DEFAULT.loads
tftp-server flash:TERM71.DEFAULT.loads
tftp-server flash:P00308000300.bin
tftp-server flash:P00308000300.loads
tftp-server flash:P00308000300.sb2
tftp-server flash:P00308000300.sbn
tftp-server flash:SCCP70.8-0-3S.loads
tftp-server flash:cvm70sccp.8-0-2-25.sbn
tftp-server flash:apps70.1-1-2-26.sbn
tftp-server flash:dsp70.1-1-2-26.sbn
tftp-server flash:cnu70.3-1-2-26.sbn
tftp-server flash:jar70sccp.8-0-2-25.sbn
radius-server host 10.13.32.241 auth-port 1645 acct-port 1646
radius-server timeout 40
radius-server deadtime 2
radius-server key cisco
radius-server vsa send accounting
!
control-plane
!
no call rsvp-sync
!
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/0:15
!
voice-port 1/1:15
!
!
!
!
!
dial-peer voice 1 voip
destination-pattern ........
voice-class codec 2
session target ras
incoming called-number 9362....
dtmf-relay h245-alphanumeric
req-qos controlled-load audio
!
dial-peer voice 2 pots
destination-pattern 93621101
!
dial-peer voice 3 pots
destination-pattern 93621102
!
dial-peer voice 10 voip
destination-pattern 2668....
voice-class codec 1
session target ipv4:10.13.46.200
!
dial-peer voice 101 voip
shutdown
destination-pattern 5694....
voice-class codec 1
session target ipv4:10.13.32.10
incoming called-number 9362....
!
dial-peer voice 102 voip
shutdown
destination-pattern 2558....
voice-class codec 1
session target ipv4:10.13.32.12
incoming called-number 9362....
!
dial-peer voice 103 voip
shutdown
destination-pattern 9845....
voice-class codec 1
session target ipv4:10.13.32.14
incoming called-number 9362....
!
dial-peer voice 104 voip
shutdown
destination-pattern 9844....
voice-class codec 1
session target ipv4:10.13.32.15
incoming called-number 9362....
!
dial-peer voice 201 pots
destination-pattern 93625...
no digit-strip
direct-inward-dial
port 1/0:15
!
dial-peer voice 202 pots
destination-pattern 93625...
no digit-strip
direct-inward-dial
port 1/1:15
!
!
gateway
timer receive-rtp 1200
!
!
!
telephony-service
load 7960-7940 P00308000300
max-ephones 4
max-dn 4
ip source-address 10.13.32.11 port 2000
auto assign 1 to 4
secure-signaling trustpoint mytrustpoint1
cnf-file location flash:
cnf-file perphone
voicemail 25589000
max-conferences 4 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
web admin system name admin password mypassword2
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern ........
tftp-server-credentials trustpoint mytrustpoint1
server-security-mode secure
device-security-mode encrypted
create cnf-files version-stamp 7960 Oct 25 2006 07:19:39
!
!
ephone-dn 1
number 93621000
name 2851-PH1
call-forward noan 25581101 timeout 10
!
!
ephone-dn 2
number 93621001
name 2851-PH2
call-forward noan 98441000 timeout 10
!
!
ephone-dn 3
number 93621002
name 2851-PH3
!
!
ephone-dn 4
number 93621003
name 2851-PH4
!
!
ephone 1
no multicast-moh
device-security-mode encrypted
mac-address 0012.4302.A7CC
type 7970
button 1:1
!
!
!
ephone 2
no multicast-moh
device-security-mode encrypted
mac-address 0017.94CA.9CCD
type 7960
button 1:2
!
!
!
ephone 3
no multicast-moh
device-security-mode encrypted
mac-address 0017.94CA.9833
type 7960
button 1:3
!
!
!
ephone 4
no multicast-moh
device-security-mode none
mac-address 0017.94CA.A141
type 7960
button 1:4
!
!
!
line con 0
logging synchronous level all limit 20480000
line aux 0
line vty 0 4
!
scheduler allocate 20000 1000
ntp clock-period 17179791
ntp server 10.13.32.12
!
webvpn context Default_context
ssl authenticate verify all
!
no inservice
!
!
end
Where to Go Next
PKI Management
Cisco IOS public key infrastructure (PKI) provides certificate management to support security protocols
such as IP Security (IPsec), secure shell (SSH), and secure socket layer (SSL). For more information,
see the following documents:
• “Part 5: Implementing and Managing a PKI” in the Cisco IOS Security Configuration Guide for your
Cisco IOS release.
• Cisco IOS Security Command Reference for your Cisco IOS release.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • “Implementing and Managing a PKI” section in the
Cisco IOS Security Configuration Guide.
• Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 22 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes automatic line selection features in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Information About Automatic Line Selection, page 461
• How to Configure Automatic Line Selection, page 462
• Configuration Examples for Automatic Line Selection, page 464
• Additional References, page 465
• Feature Information for Automatic Line Selection, page 466
The Automatic Line Selection feature allows you to specify, on a per-phone basis, the line that is selected
when you pick up a phone handset.
Any of the following behaviors can be assigned on a per-phone basis:
• Automatic line selection—Picking up the handset answers the first ringing line or, if no line is
ringing, selects the first idle line. Use the auto-line command with no keyword or argument. This is
the default.
• Manual line selection (no automatic line selection)—Pressing the Answer soft key answers the first
ringing line, and pressing a line button selects a line for an outgoing call. Picking up the handset
does not answer calls or provide dial tone. Use the no auto-line command.
• Automatic line selection for incoming calls only—Picking up the handset answers the first ringing
line, but if no line is ringing, it does not select an idle line for an outgoing call. Pressing a line button
selects a line for an outgoing call. Use the auto-line incoming command.
• Automatic line selection for outgoing calls only—Picking up the handset for an outgoing call selects
the line associated with the button-number argument. If a button number is specified and the line
associated with that button is unavailable (because it is a shared line in use on another phone), no
dial tone is heard when the handset is lifted. You must press an available line button to make an
outgoing call. Incoming calls must be answered by pressing the Answer soft key or pressing a
ringing line button. Use the auto-line command with the button-number argument.
• Automatic line selection for incoming and outgoing calls—Pressing the Answer soft key or picking
up the handset answers an incoming call on the line associated with the specified button. Picking up
the handset for outgoing calls selects the line associated with the specified button. Use the auto-line
command with the button-number argument and answer-incoming keyword.
Restrictions
Automatic line selection is bypassed if it is configured for a trunk directory number and the line is seized
by pressing the Park or Callfwd soft keys. The first available directory number is seized.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. auto-line [button-number [answer-incoming] | incoming]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number for the phone on
Example: which you want to configure automatic line selection.
Router(config)# ephone 24
Step 4 auto-line [button-number [answer-incoming] | Assigns a type of line selection behavior to this phone.
incoming]
• auto-line—Picking up the handset answers the first
ringing line or, if no line is ringing, selects the first idle
Example:
Router(config-ephone)# auto-line 5 line. This is the default.
answer-incoming • auto-line button-number—Picking up the handset for
an outgoing call selects the line associated with the
specified button. The default if this argument is not
used is the topmost available line.
• auto-line button-number answer-incoming—Picking
up the handset answers the incoming call on the line
associated with the specified button.
• auto-line incoming—Picking up the handset answers
the first ringing line but, if no line is ringing, does not
select an idle line for an outgoing call. Pressing a line
button selects a line for an outgoing call.
• no auto-line—Disables automatic line selection.
Pressing the Answer soft key answers the first ringing
line, and pressing a line button selects a line for an
outgoing call. Picking up the handset does not answer
calls or provide dial tone.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
ephone 2
headset auto-answer line 1
headset auto-answer line 4
ephone-template 1
mac-address 011F.9010.1790
paging-dn 48
type 7960
no dnd feature-ring
no auto-line
Step 2 Use the show telephony-service ephone command to display only ephone configuration information.
Router# show telephony-service ephone
ephone 4
device-security-mode none
username "Accounting"
mac-address FF0E.4857.5E91
button 1c34,35
no auto-line
The following example enables automatic selection of line button 1 when the handset is lifted to answer
incoming calls or to make outgoing calls.
ephone 1
mac-address 0001.0002.0003
type 7960
auto-line 1 answer-incoming
button 1:1 2:2 3:3
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 23 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
.
Table 23 Feature Information for Automatic Line Selection
This chapter describes Call Blocking features in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Call Blocking, page 467
• How to Configure Call Blocking, page 469
• Configuration Examples for Call Blocking, page 479
• Where to Go Next, page 481
• Additional References, page 481
• Feature Information for Call Blocking, page 483
When PINs are configured for call-blocking override, they are cleared at a specific time of day or after
phones have been idle for a specific amount of time. The time of day and amount of time can be set by
the system administrator, or the defaults can be accepted.
For configuration information, see the following sections:
• “Configuring Call Blocking Exemption for a Dial Peer” section on page 475.
• “SCCP: Configuring Call Blocking Exemption for an Individual Phone” section on page 476.
• “SIP: Configuring Call Blocking Exemption for an Individual Phone or Directory Number” section
on page 477.
Class of Restriction
Class of restriction (COR) is the capability to deny certain call attempts based on the incoming and
outgoing class of restrictions provisioned on the dial peers. COR specifies which incoming dial peer can
use which outgoing dial peer to make a call. Each dial peer can be provisioned with an incoming and an
outgoing COR list.
COR functionality provides flexibility in network design by allowing users to block calls (for example,
calls to 900 numbers) and allowing different restrictions to call attempts from different originators.
For configuration information, see the “SCCP: Applying Class of Restriction to a Directory Number”
section on page 470.
Prerequisites
• COR lists must be created in dial peers. For information, see the “Class of Restrictions” section in
the “Dial Peer Configuration on Voice Gateway Routers” document in the Cisco IOS Voice
Configuration Library.
• Directory number to which COR is to be applied must be configured in Cisco Unified CME. For
configuration information, see “SCCP: Creating Directory Numbers” on page 158.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. corlist {incoming | outgoing} cor-list-name
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
Example:
Router(config)# ephone-dn 12
Step 4 corlist {incoming | outgoing} Configures a COR on the dial peers associated with an ephone-dn.
cor-list-name
Example:
Router(config-ephone-dn)# corlist
outgoing localcor
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
Prerequisites
• Cisco unified CME 3.4 or a later version.
• COR lists must be created in dial peers. For information, see the “Class of Restrictions” section in
the “Dial Peer Configuration on Voice Gateway Routers” document in the Cisco IOS Voice
Configuration Library.
• Individual phones to which COR is to be applied must be configured in Cisco Unified CME. For
configuration information, see “SIP: Creating Directory Numbers” on page 162.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] |
default}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 cor { incoming | outgoing} cor-list-name Configures a class of restriction (COR) for the dynamically
{cor-list-number starting-number [- created VoIP dial peers associated with directory numbers
ending-number] | default}
and specifies which incoming dial peer can use which
outgoing dial peer to make a call.
Example: • Each dial peer can be provisioned with an incoming and
Router(config-register-pool)# cor incoming
an outgoing COR list.
call91 1 91011
ephone-dn 23
number 2835
corlist outgoing 5x
Step 2 Use the show dialplan dialpeer command to determine which outbound dial peer is matched for an
incoming call, based on the COR criteria and the dialed number specified in the command line. Use the
timeout keyword to enable matching variable-length destination patters associated with dial peers. This
can increase your chances of finding a match for the dial peer number you specify.
Router# show dialplan dialpeer 300 number 1900111
VoiceOverIpPeer900
information type = voice,
description = `',
tag = 900, destination-pattern = `1900',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 900, Admin state is up, Operation state is up,
Step 3 Use the show dial-peer voice command to display the attributes associated with a particular dial peer.
Router# show dial-peer voice 100
VoiceEncapPeer100
information type = voice,
description = `',
tag = 100, destination-pattern = `',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 100, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `555....', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
huntstop = disabled,
in bound application associated: 'vxml_inb_app'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
type = pots, prefix = `',
forward-digits default
session-target = `', voice-port = `',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with GK = TRUE
Restrictions
• Before Cisco CME 3.3, call blocking is not supported on analog phones connected to Cisco ATAs
or FXS ports in H.323 mode.
• Before Cisco CME 3.4, call blocking is not supported on SIP IP phones connected directly in
Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. after-hours block pattern tag pattern [7-24]
5. after-hours day day start-time stop-time
6. after-hours date month date start-time stop-time
7. login [timeout [minutes]] [clear time]
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony service Enters telephony service configuration mode.
Example:
Router(config)# telephony service
Step 4 after-hours block pattern pattern-tag pattern Defines pattern to be matched for blocking calls from IP
[7-24] phones.
• pattern-tag—Unique number pattern for call blocking.
Example: Define up to 32 call-blocking patterns in separate
Router(config-telephony)# after-hours block commands. Range is 1 to 32.
pattern 2 91
Example:
Router(config-telephony)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag {pots | voatm | vofr | voip}
4. paramspace callsetup after-hours-exempt true
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag {pots | voatm | vofr | Defines a particular dial peer, specifies the method of voice
voip} encapsulation, and enters dial-peer configuration mode.
Example:
Router(config)# dial peer voice 501 voip
Step 4 paramspace callsetup after-hours-exempt true Exempts a dial peer from call blocking configuration.
Example:
Router(config-dialpeer)# paramspace callsetup
after-hours-exempt true
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-dialpeer)# end
or
Router(config-register-dn)# end
Restrictions
Call blocking override is supported only on phones that support soft-key display.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. after-hour exempt
5. pin pin-number
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—The unique sequence number for the phone
Example: that is to be exempt from call blocking.
Router(config)# ephone 4
Step 4 after-hour exempt Specifies that this phone is exempt from call blocking.
Phones exempted in this manner are not restricted from any
call-blocking patterns and no authentication of the phone
Example:
Router(config-ephone)# after-hour exempt
user is required.
Step 5 pin pin-number Declares a personal identification number (PIN) that is used
to log into an ephone.
Example: • pin-number—Number from four to eight digits in
Router(config-ephone)# pin 5555 length.
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Restrictions
The Login toll-bar override is not supported on SIP IP phones; there is no pin to bypass blocking on IP
phones that are connected to Cisco Unified CME and running SIP.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
or
voice register dn dn-tag
4. after-hour exempt
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
or parameters for specified SIP phone.
voice register dn dn-tag or
Enters voice register dn mode to define a directory number
Example: for a SIP phone, intercom line, voice port, or an MWI.
Router(config)# voice register pool 1
or
Router(config)# voice register dn 1
Step 4 after-hour exempt Exempts all numbers on a SIP phone from call blocking.
or
Example: Exempts an individual directory number from call blocking.
Router(config-register-pool)# after-hour exempt
or
Router(config-register-dn)# after-hour exempt
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end
or
Router(config-register-dn)# end
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
web admin system name sys3 password sys3
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
after-hours block pattern 1 91900 7-24
after-hours block pattern 2 9976 7-24
after-hours block pattern 3 9011 7-24
after-hours block pattern 4 91...976.... 7-24
!
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28
Step 2 Use the show ephone login command to display the login status of all phones.
Router# show ephone login
after-hour exempt
!
ephone 24
mac 2234.1543.6352
button 1:34
The following example deactivates a phone’s login after three hours of idle time and clears all logins at
10 p.m.:
ephone 1
pin 1000
!
telephony-service
login timeout 180 clear 2200
ephone-dn 1
corlist incoming user1
corlist outgoing user1
!
ephone-dn 2
corlist incoming user2
corlist outgoing user2
Where to Go Next
After modifying a configuration for a Cisco Unified IP phone connected to Cisco Unified CME, you
must reboot the phone to make the changes take effect. For more information, see “Resetting and
Restarting Phones” on page 257.
Ephone-dn Templates
The corlist command can be included in an ephone-dn template that is applied to one or more
ephone-dns. For more information, see “Creating Templates” on page 881.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 24 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the call park feature in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Call Park, page 485
• How to Configure Call Park, page 490
• Configuration Examples for Call Park, page 495
• Where to Go Next, page 496
• Additional References, page 497
• Feature Information for Call Park, page 498
You can create a call-park slot that is reserved for use by one extension by assigning that slot a number
whose last two digits are the same as the last two digits of the extension. When an extension starts to
park a call, the system searches first for a call-park slot that has the same final two digits as the extension.
If no such call-park slot exists, the system chooses an available call-park slot.
Multiple call-park slots can be created with the same extension number so that more than one call can
be parked for a particular department or group of people at a known extension number. For example, at
a hardware store, calls for the plumbing department can be parked at extension 101, calls for lighting
can be parked at 102, and so forth. Everyone in the plumbing department knows that calls parked at 101
are for them and can pick up calls from extension 101. When multiple calls are parked at the same
call-park slot number, they are picked up in the order in which they were parked; that is, the call that has
been parked the longest is the first call picked up from that call-park slot number.
If multiple call-park slots use the same extension number, you must configure each ephone-dn that uses
the extension number with the no huntstop command, except for the last ephone-dn to which calls are
sent. In addition, each ephone-dn must be configured with the preference command. The preference
numeric values must increase to match the order of the ephone-dns. That is, the lowest ephone-dn tag
park-slot must have the lowest numeric preference number, and so forth. Without the configuration of the
preference and huntstop commands, all calls that are parked after a second call has been parked will generate
a busy signal. The caller who is being transferred to park will hear a busy signal, while the phone user who
parked the call will receive no indication that the call was lost.
A reminder ring can be sent to the extension that parked the call by using the timeout keyword with the
park-slot command. The timeout keyword and argument set the interval length during which the
call-park reminder ring is timed out or inactive. If the timeout keyword is not used, no reminder ring is
sent to the extension that parked the call. The number of timeout intervals and reminder rings are
configured with the limit keyword and argument. For example, a limit of 3 timeout intervals sends 2
reminder rings (interval 1, ring 1, interval 2, ring 2, interval 3). The timeout and limit keywords and
arguments also set the maximum time that calls stay parked. For example, a timeout interval of 10
seconds and a limit of 5 timeout intervals (park-slot timeout 10 limit 5) will park calls for
approximately 50 seconds.
The reminder ring is sent only to the extension that parked the call unless the notify keyword is also used
to specify an additional extension number to receive a reminder ring. When an additional extension
number is specified using the notify keyword, the phone user at that extension can retrieve a call from
this slot by pressing the PickUp soft key and the asterisk (*) key.
You can define both the length of the timeout interval for calls parked at a call-park slot and the number
of timeout intervals that should occur before the call is either recalled or transferred. If you specify a
transfer target in the park-slot command, the call is transferred to the specified target after the timeout
intervals expire rather than to the primary number of the parking phone.
If a name has been specified for the call-park slot using the name command, that name will be displayed
on a recall or transfer rather than an extension number.
You can also specify an alternate target extension to which to transfer a parked call if the recall or
transfer target is in use. In use is defined as either ringing or connected. For example, a call is parked at
the private park slot for the phone with the primary extension of 2001, as shown in Figure 23. After the
timeouts expire, the system attempts to recall the call to extension 2001, but that line is connected to
another call. The system then transfers the call to the alternate target, extension 3784.
2001 2 3754
2002
2003 1
Dedicated
Call-Park Slot 3
3333
1. A user on extension 2003
ephone-dn 1 parks a call using the Park
number 2001
soft key.
ephone-dn 2 2. After three intervals of 60
number 2002
seconds, the call is recalled to the
ephone-dn 3 phone’s primary number, 2001.
number 2003 3. If 2001 is busy, the call is
ephone-dn 4 transferred to 3754.
number 3333
name Park 2001
135130
park-slot reserved-for 2001 timeout 60 limit 3 recall alternate 3754
ephone 2
button 1:1 2:2 3:3
Call-Park Blocking
In Cisco Unified CME 4.0 and later versions, individual ephones can be prevented from making transfers
to call-park slots by using the transfer-park blocked command. This command prevents transfers to
park that use the Transfer soft key and a call-park slot number, while allowing call-parks that use only
the Park soft key. (To prevent use of the Park soft key, use an ephone template to remove it from the
phone. See “Customizing Soft Keys” on page 829.)
An exception is made for phones with reserved, or dedicated, park slots. If the transfer-park blocked
command is used on an ephone that has a dedicated park slot, the phone is blocked from parking calls at
park slots other than the phone’s dedicated park slot but can still park calls at its own dedicated park slot.
Call-Park Redirect
By default, H.323 and SIP calls that use the call-park feature use hairpin call forwarding or transfer to
park calls and to pick up calls from park. The call-park system redirect command allows you to specify
that these calls should use H.450 or the SIP Refer method of call forwarding or transfer. The no form of
the command returns the system to the default behavior.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. park-slot [reserved-for extension-number] [timeout seconds limit count] [notify
extension-number [only]] [recall] [transfer extension-number] [alternate extension-number]
[retry seconds limit count]
6. exit
7. ephone phone-tag
8. transfer-park blocked
9. exit
10. telephony-service
11. call-park system redirect
12. restart all
13. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 20 ephone-dn during configuration tasks. The maximum number
of ephone-dns for a particular Cisco Unified CME system is
version- and platform-specific.
• dual-line—(Optional) Enables an ephone-dn with one voice
port and two voice channels, which supports features such as
call waiting, call transfer, and conferencing with a single
ephone-dn.
Step 4 number number [secondary number] [no-reg Configures a valid extension number for this ephone-dn instance.
[both | primary]]
• number—String of up to 16 digits that represents a telephone
or extension number to be associated with this ephone-dn.
Example:
Router(config-ephone-dn)# number 2345
• secondary—(Optional) Allows you to associate a second
telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should not
register with the H.323 gatekeeper. Unless you specify one of
the optional keywords (both or primary) after the no-reg
keyword, only the secondary number is not registered.
Example:
Router(config-ephone-dn)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 25
Step 8 transfer-park blocked (Optional) Prevents extensions on this ephone from transferring
calls to call-park slots.
Example: Note This command prevents the use of the Transfer soft key
Router(config-ephone)# transfer-park and slot number to transfer calls to park slots. It does not
blocked prevent use of the Park soft key.
Step 9 exit Exits ephone configuration mode.
Example:
Router(config-telephony)# exit
Step 10 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 11 call-park system redirect Specifies that within the call-park feature, H.323 and SIP calls
will use H.450 or the SIP Refer method of call forwarding or
transfer to park calls and to pick up calls from park.
Example:
Router(config)# call-park system redirect The no form of the command returns to the default behavior,
which is to use hairpin call forwarding or transfer to park calls and
pick up calls from park.
Step 12 restart all Performs a fast reboot of all phones associated with this
Cisco Unified CME router. Does not contact the DHCP server.
Example: Note The first time that call-park slots are defined, IP phones
Router(config)# restart all must be rebooted before the Park soft key is displayed on
phones. This command is not required after subsequent
call-park slot definitions.
Step 13 exit Returns to privileged EXEC mode.
Example:
Router(config)# exit
!
ephone-dn 23
number 853
park-slot timeout 10 limit 1 recall
description park slot for Sales
!
!
ephone-dn 24
number 8126
park-slot reserved-for 126 timeout 10 limit 1 transfer 8145
!
!
ephone-dn 25
number 8121 secondary 121
park-slot reserved-for 121 timeout 30 limit 1 transfer 8145
!
!
ephone-dn 26
number 8136 secondary 136
park-slot reserved-for 136 timeout 10 limit 1 recall
!
!
ephone-dn 30 dual-line
number 451 secondary 501
preference 10
huntstop channel
!
!
ephone-dn 31 dual-line
number 452 secondary 502
preference 10
huntstop channel
!
Step 2 Use the show telephony-service ephone-dn command to display call park configuration information.
Router# show telephony-service ephone-dn
ephone-dn 26
number 8136 secondary 136
park-slot reserved-for 136 timeout 10 limit 1 recall
Step 2 Use the debug ephone commands to observe messages and states associated with an ephone. For more
information, see the Cisco Unified CME Command Reference.
ephone-dn 12
number 235
ephone-dn 13
number 236
ephone 25
button 1:11 2:12 3:13
transfer-park blocked
The following example sets up a dedicated park slot for the extensions on ephone 6 and blocks transfers
to call park from extensions 2977, 2978, and 2979 on that phone. Those extensions can still park calls
at the phone’s dedicated park slot by using the Park soft key or the Transfer soft key and the FAC for call
park.
ephone-dn 3
number 2558
name Park 2977
park-slot reserved-for 2977 timeout 60 limit 3 recall alternate 3754
ephone-dn 4
number 2977
ephone-dn 5
number 2978
ephone-dn 6
number 2979
ephone 6
button 1:4 2:5 3:6
transfer-park blocked
Where to Go Next
Controlling Use of the Park Soft Key
To block the functioning of the call park (Park) soft key without removing the key display, create and
apply an ephone template that contains the features blocked command. For more information, see
“Customizing Soft Keys” on page 829.
To remove the call park (Park) soft key from one or more phones, create and apply an ephone template
that contains the appropriate softkeys command. For more information, see “Customizing Soft Keys”
on page 829.
Ephone Templates
The transfer-park blocked command, which blocks transfers to call-park slots, can be included in
ephone templates that are applied to individual ephones.
The Park soft key can be removed from the display of one or more phones by including the appropriate
softkeys command in an ephone template and applying the template to individual ephones.
For more information, see “Creating Templates” on page 881.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 25 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes call transfer and forwarding features in Cisco Unified Communications Manager
Express (Cisco Unified CME) to enable interworking with various network requirements.
Contents
• Information About Call Transfer and Forwarding, page 499
• How to Configure Call Transfer and Forwarding, page 518
• Configuration Examples for Call Transfer and Forwarding, page 552
• Where to Go Next, page 559
• Additional References, page 560
• Feature Information for Call Transfer and Forwarding, page 561
Call Forwarding
Call forwarding diverts calls to a specified number under one or more of the following conditions:
• All calls—When all-call call forwarding is activated by a phone user, all incoming calls are diverted.
The target destination for diverted calls can be specified in the router configuration or by the phone
user with a soft key or feature access code. The most recently entered destination is recognized by
Cisco Unified CME, regardless of how it was entered.
• No answer—Incoming calls are diverted when the extension does not answer before the timeout
expires. The target destination for diverted calls is specified in the router configuration.
• Busy—Incoming calls are diverted when the extension is busy and call waiting is not active. The
target destination for diverted calls is specified in the router configuration.
• Night service—All incoming calls are automatically diverted during night-service hours. The target
destination for diverted calls is specified in the router configuration.
A directory number can have all four types of call forwarding defined at the same time with a different
forwarding destination defined for each type of call forwarding. If more than one type of call forwarding
is active at one time, the order for evaluating the different types is as follows:
1. Call forward night-service
2. Call forward all
3. Call forward busy and call forward no-answer
H.450.3 capabilities are enabled globally on the router by default, and can be disabled either globally or
for individual dial peers. You can configure incoming patterns for using the H.450.3 standard.
Calling-party numbers that do not match the patterns defined with this command are forwarded using
Cisco-proprietary call forwarding for backward compatibility. For information about configuring
H.450.3 on a Cisco Unified CME system, see the “SCCP: Enabling Call Forwarding for a Directory
Number” section on page 523.
• A POTS dial peer for the secondary number as expanded by the dialplan-pattern command
Call forwarding is normally applied to all dial peers created for an ephone-dn. Selective call forwarding
allows you to apply call forwarding for busy or no-answer calls only for the dial peers you have specified,
based on the called number that was used to route the call to the ephone-dn.
For example, the following commands set up a single ephone-dn (ephone-dn 5) with four dial peers:
telephony-service
dialplan-pattern 1 40855501.. extension-length 4 extension-pattern 50..
ephone-dn 5
number 5066 secondary 5067
In this example, selective call forwarding can be applied so that calls are forwarded when:
• callers dial the primary number 5066.
• when callers dial the secondary number 5067.
• when callers dial the expanded numbers 4085550166 or 4085550167.
For configuration information, see the “SCCP: Enabling Call Forwarding for a Directory Number”
section on page 523.
In Cisco Unified CME 4.1 and later, the following enhancements are supported for the
Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE to keep the
configuration consistent between Cisco Unified CME and the SIP phone:
• When Call Forward All is configured on Cisco Unified CME with the call-forward b2bua all
command, the configuration is sent to the phone which updates the CfwdAll soft key to indicate that
Call forward All is enabled. Because Call Forward All is configured on a per line basis, the CfwdAll
soft key is updated only when Call Forward All is enabled for the primary line.
• When a user enables Call Forward All on a phone using the CfwdAll soft key, the uniform resource
identifier (URI) for the service (defined with the call-feature-uri command) and the call forward
number (unless Call Forward All is disabled) is sent to Cisco Unified CME. It updates its voice
register pool and voice register dn configuration with the call-forward b2bua all command to be
consistent with the phone configuration.
• Call Forward All supports KPML so that a user does not need to press the Dial or # key, or wait for
the interdigit timeout, to configure the Call Forward All number. Cisco Unified CME collects the
Call Forward All digits until it finds a match in the dial peers.
For configuration information, see the “SIP: Configuring Call-Forwarding-All Soft Key URI” section
on page 548.
Call Transfer
When you are connected to another party, call transfer allows you to shift the connection of the other
party to a different number. Call transfer methods must interoperate with systems in the other networks
with which you interface. Cisco CME 3.2 and later versions provide full call-transfer and
call-forwarding interoperability with call processing systems that support H.450.2, H.450.3, and
H.450.12 standards. For call processing systems that do not support H.450 standards, Cisco CME 3.2
and later versions provide VoIP-to-VoIP hairpin call routing.
Call transfers can be blind or consultative. A blind transfer is one in which the transferring extension
connects the caller to a destination extension before ringback begins. A consultative transfer is one in
which the transferring party either connects the caller to a ringing phone (ringback heard) or speaks with
the third party before connecting the caller to the third party.
You can configure blind or consultative transfer on a systemwide basis or for individual extensions. For
example, in a system that is set up for consultative transfer, a specific extension with an auto-attendant
that automatically transfers incoming calls to specific extension numbers can be set to use blind transfer,
because auto-attendants do not use consultative transfer.
Direct Station Select consultative transfer is enabled with the transfer-system full-consult dss
command, which defines the call transfer method for all lines served by the router. The transfer-system
full-consult dss command supports the keep-conference command. See “Configuring Conferencing”
on page 647.
• Cisco Unified CME supports all SIP Refer method call transfer scenarios, but you must ensure that
call transfer is enabled using H.450.2 standards.
• H.450 standards are not supported by Cisco Unified Communications Manager, Cisco BTS, or
Cisco PGW, although hairpin call routing or an H.450 tandem gateway can be set up to handle calls
to and from those types of systems.
The following series of figures depicts a call being transferred using H.450.2 standards. Figure 24 on
page 504 shows A calling B. Figure 25 on page 504 shows B consulting with C and putting A on hold.
Figure 26 on page 505 shows that B has connected A and C, and Figure 27 on page 505 shows A and C
directly connected, with B no longer involved in the call.
H.323
V
Media Termination
Cisco Unified CME 1 Point (MTP)
IP
Phone A Phone C
146629
IP
Phone B
H.323 V
Non-H.450
Cisco Unified CME 1 gateway
IP
Cisco Unified CME 2 Phone C
Phone A
IP
Phone B
H.323 V
Non-H.450
Unified
Cisco CME 1 CME 1 gateway
IP
Phone A Cisco Unified CME 2 Phone C
146633
IP
Phone B
H.323 V
Non-H.450
Cisco Unified CME 1 gateway
IP
Cisco Unified CME 2 Phone C
Phone A
146634
IP
Phone B
H.450.12 Support
Cisco CME 3.1 and later versions support the H.450.12 call capabilities standard, which provides a
means to advertise and dynamically discover H.450.2 and H.450.3 capabilities in voice gateway
endpoints on a call-by-call basis. When discovered, the calls associated with non-H.450 endpoints can
be directed to use non-H.450 methods for transfer and forwarding, such as hairpin call routing or H.450
tandem gateway.
When H.450.12 is enabled, H.450.2 and H.450.3 services are disabled for call transfers and call forwards
unless a positive H.450.12 indication is received from all other VoIP endpoints involved in the call. If a
positive H.450.12 indication is received, the router uses the H.450.2 standard for call transfers and the
H.450.3 standard for call forwarding. If a positive H.450.12 indication is not received, the router uses
the alternative method that you have configured for call transfers and forwards, either hairpin call routing
or an H.450 tandem gateway.
You can have either of the following situations in your network:
• All gateway endpoints support H.450.2 and H.450.3 standards. In this situation, no special
configuration is required because support for H.450.2 and H.450.3 standards is enabled on the
Cisco CME 3.1 or later router by default. H.450.12 capability is disabled by default, but it is not
required because all calls can use H.450.2 and H.450.3 standards.
• Not all gateway endpoints support H.450.2 and H.450.3 standards. Therefore, specify how
non-H.450 calls are to be handled by choosing one of the following options:
– Enable the H.450.12 capability in Cisco CME 3.1 and later to dynamically determine, on a
call-by-call basis, whether each call has H.450.2 and H.450.3 support. If H.450.12 is enabled
and a call is determined to have H.450 support, the call is transferred using H.450.2 standards
or forwarded using H.450.3 standards. See the “Enabling H.450.12 Capabilities” section on
page 529.
Support for the H.450.12 standard is disabled by default and can be enabled globally or for
individual dial peers.
If the call does not have H.450 support, it can be handled by a VoIP-to-VoIP connection that you
configure using dial peers and the “Enabling H.323-to-H.323 Connection Capabilities” section
on page 531. The connection can be used for hairpin call routing or routing to an H.450 tandem
gateway.
– Explicitly disable H.450.2 and H.450.3 capability on a global basis or by individual dial peer,
which forces all calls to be handled by a VoIP-to-VoIP connection that you configure using dial
peers and the“Enabling H.323-to-H.323 Connection Capabilities” section on page 531. This
connection can be used for hairpin call routing or routing to an H.450 tandem gateway.
In Cisco CME 3.2 and later versions, transcoding between G.711 and G.729 is supported when one leg
of a VoIP-to-VoIP hairpin call uses G.711 and the other leg uses G.729. For information about
transcoding, see “Configuring Transcoding Resources” on page 303.
Hairpin call routing provides the following benefits:
• Call transfer and forwarding is provided to non-H.450 endpoints, such as
Cisco Unified Communications Manager, Cisco BTS, or Cisco PGW.
• The network can also contain Cisco CME 3.0 or Cisco ITS 2.1 systems.
Hairpin call routing has the following disadvantages:
• End-to-end signaling and media delay are increased significantly.
• A single hairpinned call uses as much WAN bandwidth as two directly connected calls.
VoIP-to-VoIP hairpin connections can be made using dial peers if the allow-connections h323 to h323
command is enabled and at least one of the following is true:
• H.450.12 is used to detect calls on which H.450.2 or H.450.3 is not supported by the remote system.
• H.450.2 or H.450.3 is explicitly disabled.
• Cisco Unified CME automatically detects that the remote system is a
Cisco Unified Communications Manager.
Figure 28 on page 508 shows a call that is made from A to B. Figure 29 on page 509 shows that B has
forwarded all calls to C. Figure 30 on page 509 shows that A and C are connected by an H.323 hairpin.
H.323
V
Media Termination
Cisco Unified CME 1 Point (MTP)
IP
Phone A Phone C
IP
Phone B
H.323 V
Non-H.450
Cisco Unified CME 1 gateway
IP
Phone A Phone C
Cisco Unified CME 2
146630
IP Calls are forwarded
Phone B to phone C
H.323 V
Non-H.450
Cisco Unified CME 1 gateway
IP
Phone A Phone C
146631
IP
Phone B
Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally. For
configuration information, see the “Enabling H.323-to-H.323 Connection Capabilities” section on
page 531.
Note An H.450 tandem gateway that is used in a network to support non-H.450-capable call processing
systems requires the Integrated Voice and Video Services feature license. This feature license, which was
introduced in March 2004, includes functionality for H.323 gatekeeper, IP-to-IP Gateway, and H.450
tandem gateway. With Cisco IOS Release 12.3(7)T, an H.323 gatekeeper feature license is required with a
JSX Cisco IOS image on the selected router. Consult your Cisco Unified CME SE regarding the required
feature license. With Cisco IOS Release 12.3(7)T, you cannot use Cisco Unified CME and H.450 tandem
gateway functionality on the same router.
VoIP-to-VoIP connections can be made for an H.450 tandem gateway if the allow-connections h323 to
h323 command is enabled and one or more of the following is true:
• H.450.12 is used to dynamically detect calls on which H.450.2 or H.450.3 is not supported by the
remote VoIP system.
• H.450.2 or H.450.3 is explicitly disabled.
• Cisco CME 3.1 or later automatically detects that the remote system is a
Cisco Unified Communications Manager.
For Cisco CME 3.1 and earlier, the only type of VoIP-to-VoIP connection supported by
Cisco Unified CME is H.323-to-H.323. For Cisco CME 3.2 and later versions, H.323-to-SIP
connections are allowed only for Cisco Unified CME systems running Cisco Unity Express.
Figure 31 on page 511 shows a tandem voice gateway that is located between the central hub of the
network of a CPE-based Cisco CME 3.1 or later network and a Cisco Unified Communications Manager
network. This topology would work equally well with a Cisco BTS or Cisco PGW in place of the
Cisco Unified Communications Manager.
In the network topology in Figure 31 on page 511, the following events occur (refer to the event numbers
on the illustration):
1. A call is generated from extension 4002 on phone 2, which is connected to a
Cisco Unified Communications Manager. The H.450 tandem gateway receives the H.323 call and,
acting as the H.323 endpoint, the H.450 tandem gateway handles the call connection to a
Cisco Unified IP phone in a CPE-based Cisco CME 3.1 or later network.
2. The call is received by extension 1001 on phone 3, which is connected to Cisco Unified CME 1.
Extension 1001 performs a consultation transfer to extension 2001 on phone 5, which is connected
to Cisco Unified CME 2.
3. When extension 1001 transfers the call, the H.450 tandem gateway receives an H.450.2 message
from extension 1001.
4. The H.450 tandem gateway terminates the call leg from extension 1001 and reoriginates a call leg
to extension 2001, which is connected to Cisco Unified CME 2.
5. Extension 4002 is connected with extension 2001.
IP-to-IP
Gateway
H.450.2 Message
Private VoIP Telephone
Cisco Unified CME 2
1 Cisco Unified CME 2
V V
2 5
4
IP IP IP IP
Phone 3 Phone 4 Phone 5 Phone 6
1001 1002 3001 3002
146622
Dial Peers
Dial peers describe the virtual interfaces to or from which a call is established. All voice technologies
use dial peers to define the characteristics associated with a call leg. Attributes applied to a call leg
include specific quality of service (QoS) features, compression/decompression (codec), voice activity
detection (VAD), and fax rate. Dial peers are also used to establish the routing paths in your network,
including special routing paths such as hairpins and H.450 tandem gateways. Dial peer settings override
the global settings for call forward and call transfer. For information about configuring dial peers, see
the Dial Peer Configuration on Voice Gateway Routers guide.
IP 1001 IP 2001
IP 1002 IP 2002
IP 1003 IP 2003
QSIG 3001
3002
PBX
3003
Message
135562
center
The following QSIG supplementary service features are supported in Cisco Unified CME systems. They
follow the standards from the European Computer Manufacturers Association (ECMA) and the
International Organization for Standardization (ISO) on PRI and BRI interfaces.
• Basic calls between IP phones and PBX phones.
• Calling Line/Name Identification (CLIP/CNIP) presented on an IP phone when called by a PBX
phone; in the reverse direction, such information is provided to the called endpoint.
• Connected Line/Name Identification (COLP/CONP) information provided when a PBX phone calls
an IP phone and is connected; in the reverse direction, such information presented on an IP phone.
• Call Forward using QSIG and H.450.3 to support any combination of IP phone and PBX phone,
including an IP phone in the Cisco Unified CME system that is connected to a PBX or an IP phone
in another Cisco Unified CME system across an H.323 network.
• Call forward to the PBX message center according to the configured policy. The other two endpoints
can be a mixture of IP phone and PBX phones.
• Hairpin call transfer, which interworks with a PBX in transfer-by-join mode. Note that
Cisco Unified CME does not support the actual signaling specified for this transfer mode (including
the involved FACILITY message service APDUs) which are intended for an informative purpose
only and not for the transfer functionality itself. As a transferrer (XOR) host, Cisco Unified CME
simply hairpins two call legs to create a connection; as a transferee (XEE) or transfer-to (XTO) host,
it will not be aware of a transfer that is taking place on an existing leg. As a result, the final endpoint
may not be updated with the accurate identity of its peer. Both blind transfer and consult transfer are
supported.
• Message-waiting indicator (MWI) activation or deactivation requests are processed from the PBX
message center.
• The PBX message center can be interrogated for the MWI status of a particular ephone-dn.
• A user can retrieve voice messages from a PBX message center by making a normal call to the
message center access number.
For information about enabling QSIG supplementary services, see the “Enabling H.450.7 and QSIG
Supplementary Services at a System-Level” section on page 535 and “Enabling H.450.7 and QSIG
Supplementary Services on a Dial Peer” section on page 536.
For more information about configuring Cisco Unified CME to integrate with voice-mail systems, see
“Integrating Voice Mail” on page 355.
Disabling SIP Supplementary Services for Call Forward and Call Transfer
If a destination gateway does not support supplementary services, you can disable REFER messages for
call transfers and redirect responses for call forwarding from being sent by Cisco Unified CME or
Cisco Unified SRST.
Disabling supplementary services is supported if all endpoints use SCCP or all endpoints use SIP. It is
not supported for a mix of SCCP and SIP endpoints.
Note Cisco Communications Manager Express 3.2 (Cisco CME 3.2) and later versions provide full
call-transfer and call-forwarding with call processing systems on the network that support H.450.2,
H.450.3, and H.450.12 standards. For interoperability with call processing systems that do not support
H.450 standards, Cisco CME 3.2 and later versions provide VoIP-to-VoIP hairpin call routing without
requiring the special Tool Command Language (Tcl) script that was needed in earlier versions of
Cisco Unified CME.
Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways
In a network with Cisco CME 3.1 or later, non-H.450 gateways, and Cisco IOS gateways, the H.450.2
and H.450.3 services are provided only to calling endpoints that use H.450.12 to explicitly indicate that
they are capable of H.450.2 and H.450.3 operations. Because the Cisco BTS and Cisco PGW do not
support the H.450.12 standard, calls to and from these systems that involve call transfer or forwarding
are handled using H.323-to-H.323 hairpin call routing.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). Optionally disable H.450.2
and H.450.3 capabilities on dial peers that point to non-H.450-capable systems such as
Cisco Unified Communications Manager, Cisco BTS, or Cisco PGW. See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or for specific dial peers. See the “Enabling H.450.12 Capabilities” section on
page 529.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route calls
that do not support H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 531.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.
Note If your network contains a Cisco Unified Communications Manager, also see the instructions in the
“Enabling Interworking with Cisco Unified Communications Manager” section on page 540.
Note Cisco CME 3.0 and Cisco ITS V2.1 systems do not have H.450.12 capabilities.
In a network that contains a mix of Cisco Unified CME versions and at least one non-H.450 gateway, the
simplest configuration approach is to globally disable all H.450.2 and H.450.3 services and force
H.323-to-H.323 hairpin call routing for all transferred and forwarded calls. In this case, you would
enable H.450.12 detection capabilities globally. Alternatively, you could select to enable H.450.12
capability for specific dial peers. In this case, you would not configure H.450.12 capability globally; you
would leave it in its default disabled state.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or on specific dial peers. See the “Enabling H.450.12 Capabilities” section on
page 529.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls. See the “Enabling H.323-to-H.323 Connection Capabilities”
section on page 531.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.
Note If your network contains a Cisco Unified Communications Manager, also see the instructions in the
“Enabling Interworking with Cisco Unified Communications Manager” section on page 540.
Cisco CME 3.1 or Later, Cisco Unified Communications Manager, and Cisco IOS Gateways
In a network with Cisco CME 3.1 or later, Cisco Unified Communications Manager, and Cisco IOS
gateways, Cisco CME 3.1 and later versions support automatic detection of calls to and from
Cisco Unified Communications Manager using proprietary signaling elements that are included with the
standard H.323 message exchanges. The Cisco CME 3.1 or later system uses these detection results to
determine the H.450.2 and H.450.3 capabilities of calls rather than using H.450.12 supplementary
services capabilities exchange, which Cisco Unified Communications Manager does not support. If a
call is detected to be coming from or going to a Cisco Unified Communications Manager endpoint, the
call is treated as a non-H.450 call. All other calls in this type of network are treated as though they
support H.450 standards. Therefore, this type of network should contain only Cisco CME 3.1 or later
and Cisco Unified Communications Manager call-processing systems.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or on specific dial peers. See the “Enabling H.450.12 Capabilities” section on
page 529.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls that are detected as being to or from
Cisco Unified Communications Manager. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 531.
4. Setting up specific parameters for Cisco Unified Communications Manager. See the instructions in
the “Enabling Interworking with Cisco Unified Communications Manager” section on page 540.
5. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.
Cisco CME 3.0 or an Earlier Version, Cisco Unified Communications Manager, and Cisco IOS
Gateways
Calls between the Cisco Unified Communications Manager and the older Cisco CME 3.0 or
Cisco ITS V2.1 networks need special consideration. Because Cisco CME 3.0 and Cisco ITS V2.1
systems do not support automatic Cisco Unified Communications Manager detection and also do not
natively support H.323-to-H.323 call routing, alternative arrangements are required for these systems.
To configure call transfer and forwarding on the Cisco CME 3.0 router, you can select from the
following three options:
• Use a Tcl script to handle call transfer and forwarding by invoking Tcl-script-based H.323-to-H.323
hairpin call routing (app-h450-transfer.2.0.0.9.tcl or a later version). Enable this script on all VoIP
dial peers and also under telephony-service mode, and set the local-hairpin script parameter to 1.
See the configuration instructions in the “Configuring Call Transfer” chapter of the
Cisco CallManager Express 3.0 System Administrator Guide.
• Use a loopback-dn mechanism. See “Configuring Loopback Call Routing” on page 763.
• Configure a loopback call path using router physical voice ports.
All three options force use of H.323-to-H.323 hairpin call routing for all calls regardless of whether the
call is from a Cisco Unified Communications Manager or other H.323 endpoint (including
Cisco CME 3.1 or later).
In addition to the special considerations above, configuration of the Cisco CME 3.1 or later router for
this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 519.
2. Leaving H.450.12 capability in its default disabled state. For more information, see the “Enabling
H.450.12 Capabilities” section on page 529.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls that are detected as being to or from
Cisco Unified Communications Manager. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 531.
4. Setting up specific parameters for Cisco Unified Communications Manager. See the instructions in
the “Enabling Interworking with Cisco Unified Communications Manager” section on page 540.
5. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.
SCCP
• Enabling Call Transfer and Forwarding at System-Level, page 519 (required)
• SCCP: Enabling Call Forwarding for a Directory Number, page 523 (required)
• SCCP: Enabling Call Transfer for a Directory Number, page 526 (required)
• SCCP: Configuring Call Transfer Options for Phones, page 527 (optional))
• SCCP: Verifying Call Transfer, page 528 (optional)
• Enabling H.450.12 Capabilities, page 529 (optional)
• Enabling H.323-to-H.323 Connection Capabilities, page 531 (optional)
• Forwarding Calls Using Local Hairpin Routing, page 533 (optional)
• Enabling H.450.7 and QSIG Supplementary Services at a System-Level, page 535 (optional)
• Enabling H.450.7 and QSIG Supplementary Services on a Dial Peer, page 536 (optional)
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 538 (optional)
• Enabling Interworking with Cisco Unified Communications Manager, page 540 (optional)
SIP B2BUA
• SIP: Configuring SIP-to-SIP Phone Call Forwarding, page 546 (required)
• SIP: Configuring Call-Forwarding-All Soft Key URI, page 548 (optional)
• SIP: Specifying Number of 3XX Responses To be Handled, page 549 (optional)
• SIP: Configuring Call Transfer, page 550 (required)
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 538 (optional)
Note H.450.2 and H.450.3 capabilities are enabled by default for transferred or forwarded parties and
transfer-destination or forward-destination parties. Dial peer settings override the global setting.
Prerequisites
Cisco CME 3.0 or a later version, or Cisco ITS V2.1.
Restrictions
• Call transfers are handled differently depending on the Cisco Unified CME version. See Table 26
on page 506 for recommendations on selecting a transfer method for your Cisco Unified CME
version.
• The transfer-system local-consult command is not supported if the transfer-to destination is on the
Cisco ATA, Cisco VG224, or a SCCP-controlled FXS port.
• The H.450.2 and H.450.3 standards are not supported by Cisco Unified Communications Manager,
Cisco BTS, or Cisco PGW.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. transfer-system {blind | full-blind | full-consult [dss] | local-consult}
5. transfer-pattern transfer-pattern [blind]
6. call-forward pattern pattern
7. exit
8. voice service voip
9. supplementary-service h450.2
10. supplementary-service h450.3
11. exit
12. dial-peer voice tag voip
13. supplementary-service h450.2
14. supplementary-service h450.3
15. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 transfer-system {blind | full-blind | Specifies the call transfer method.
full-consult [dss] | local-consult}
• Cisco CME 3.0 and later versions—Use only the full-blind
or full-consult keyword.
Example:
Router(config-telephony)# transfer-system
• Before Cisco CME 3.0—Use the local-consult or blind
full-consult keyword. (Cisco ITS 2.1 can use the full-blind or
full-consult keyword by also using the Tcl script in the file
called app-h450-transfer.x.x.x.x.zip.)
• blind—Calls are transferred without consultation with a
single phone line using the Cisco proprietary method. This is
the default in Cisco CME versions earlier than 4.0.
• full-blind—Calls are transferred without consultation using
H.450.2 standard methods.
• full-consult—Calls are transferred with consultation using
H.450.2 standard methods and a second phone line if
available. Calls fall back to full-blind if the second line is
unavailable. This is the default in Cisco Unified CME 4.0 and
later versions.
• dss—(Optional) Calls are transferred with consultation to
idle monitored lines. All other call-transfer behavior is
identical to full-consult.
• local-consult—Calls are transferred with local consultation
using a second phone line if available. The calls fall back to
blind for nonlocal consultation or nonlocal transfer target.
Not supported if transfer-to destination is on the Cisco ATA,
Cisco VG224, or a SCCP-controlled FXS port.
Example:
Router(config-telephony)# exit
Step 8 voice service voip (Optional) Enters voice-service configuration mode to establish
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Example:
Router(conf-voi-serv)# exit
Step 12 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 1 voip
Step 13 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services capabilities
for an individual dial peer.
Example: • Default is enabled. You can also use this command in
Router(config-dial-peer)# no voice-service configuration mode to enable H.450.2 services
supplementary-service h450.2 globally.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for the
dial peer.
Example:
Router(config-dial-peer)# end
Note When defining call forwarding to nonlocal numbers, it is important to note that pattern digit matching is
performed before translation-rule operations. Therefore, you should specify in this command the digits
actually entered by phone users before they are translated. For more information, see the “Voice
Translation Rules and Profiles” section in “Configuring Dialing Plans” on page 267.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. exit
6. ephone-dn dn-tag [dual-line]
7. number number [secondary number] [no-reg [both | primary]]
8. call-forward all target-number
9. call-forward busy target-number [primary | secondary] [dialplan-pattern]
10. call-forward noan target-number timeout seconds [primary | secondary] [dialplan-pattern]
11. call-forward night-service target-number
12. call-forward max-length length
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)#
Step 4 call-forward pattern pattern Specifies the H.450.3 standard for call forwarding. Calling-party
numbers that do not match the patterns defined with this
command are forwarded using Cisco-proprietary call forwarding
Example:
Router(config-telephony)# call-forward
for backward compatibility.
pattern .T • pattern—Digits to match for call forwarding using the
H.450.3 standard. If an incoming calling-party number
matches the pattern, it is forwarded using the H.450.3
standard. A pattern of .T forwards all calling parties using the
H.450.3 standard.
Step 5 exit Exits telephony-service configuration mode.
Example:
Router(config-telephony)# exit
Step 6 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example: • dual-line—(Optional) Enables an ephone-dn with one voice
Router(config)# ephone-dn 20 port and two voice channels, which supports features such as
call waiting, call transfer, and conferencing with a single
ephone-dn.
Step 7 number number [secondary number] [no-reg Configures a valid extension number for this ephone-dn instance.
[both | primary]]
Example:
Router(config-ephone-dn)# number 2777
secondary 2778
Example:
Router(config-ephone-dn)# call-forward
busy 2513
Step 10 call-forward noan target-number timeout Forwards calls for an extension that does not answer.
seconds [primary | secondary]
[dialplan-pattern]
Example:
Router(config-ephone-dn)# call-forward
noan 2513 timeout 45
Step 11 call-forward night-service target-number Automatically forwards incoming calls to the specified number
when night service is active.
Example: • target-number—Phone number to which calls are forwarded.
Router(config-ephone-dn)# call-forward
night-service 2879
Note Night service must also be configured. See “Configuring
Call-Coverage Features” on page 563.
Step 12 call-forward max-length length (Optional) Limits the number of digits that can be entered for a
target number when using the CfwdAll soft key on an IP phone.
Example: • length—Number of digits that can be entered using the
Router(config-ephone-dn)# call-forward CfwdAll soft key on an IP phone.
max-length 5
Step 13 no forward local-calls (Optional) Specifies that local calls (calls from ephone-dns on the
same Cisco Unified CME system) will not be forwarded from this
extension.
Example:
Router(config-ephone-dn)# no forward • If this extension is busy, an internal caller hears a busy signal.
local-calls
• If this extension does not answer, the internal caller hears
ringback.
Step 14 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
Prerequisites
Call transfer must be enabled globally. See the “Enabling Call Transfer and Forwarding at
System-Level” section on page 519.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. transfer-mode {blind | consult}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example: • dual-line—(Optional) Enables an ephone-dn with one voice
Router(config)# ephone-dn 20 port and two voice channels, which supports features such as
call waiting, call transfer, and conferencing with a single
ephone-dn.
Step 4 transfer-mode {blind | consult} Specifies the type of call transfer for an individual directory
number using the H.450.2 standard, allowing you to override the
global setting.
Example:
Router(config-ephone-dn)# transfer-mode • Default: system-level value set with the transfer-system
blind command.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Restrictions
• Transfers made to speed-dial numbers are not blocked when the transfer-pattern blocked
command is used.
• Transfers made using speed-dial are not blocked by the after-hours block pattern command.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-template template-tag
4. transfer-pattern blocked
5. transfer max-length digit-length
6. exit
7. ephone phone-tag
8. ephone-template template-tag
9. restart
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique sequence number that identifies this
Example: template during configuration tasks. Range is 1 to 20.
Router(config)# ephone-template 1
Step 4 transfer-pattern blocked (Optional) Prevents directory numbers on the phone to which this
template is applied from transferring calls to patterns specified in
the transfer-pattern (telephony-service) command.
Example:
Router(config-ephone-template)# Note This command is also available in ephone configuration
transfer-pattern blocked mode to block external transfers from individual phones
without using a template.
Example:
Router(config-ephone-template)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 25
Step 8 ephone-template template-tag Applies a template to a phone.
• template-tag—Template number that you want to apply to
Example: this phone.
Router(config-ephone)# ephone-template 1
Step 9 restart Performs a fast reboot of this phone without contacting the DHCP
server for updated information.
Example: • Repeat Step 6 to Step 9 for each phone on which you want to
Router(config-ephone)# restart limit transfer capabilities.
Step 10 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Step 2 If you have used the transfer-mode command to override the global transfer mode for an individual
ephone-dn, use the show running-config or show telephony-service ephone-dn command to verify that
setting.
Router# show running-config
!
ephone-dn 40 dual-line
number 451
description Main Number
huntstop channel
no huntstop
transfer-mode blind
Step 3 Use the show telephony-service ephone-template command to view ephone-template configurations.
Restrictions
Cisco CME 3.0 and earlier versions do not support H.450.12.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. supplementary-service h450.12 [advertise-only]
5. exit
6. dial-peer voice tag voip
7. supplementary-service h450.12
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip (Optional) Enters voice service configuration mode to establish
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 supplementary-service h450.12 (Optional) Enables H.450.12 supplementary services capabilities
[advertise-only] globally for VoIP endpoints.
• This command enables call-by-call detection of H.450
Example: capabilities when some endpoints in your mixed network are
Router(conf-voi-serv)# H.450-capable and other endpoints are not. This command is
supplementary-service h450.12
disabled by default.
• advertise-only—(Optional) Advertises H.450 capabilities to
the remote end but does not require H.450.12 responses. Use
this keyword on Cisco CME 3.1 or later systems if you have
a mixed network containing Cisco CME 3.0 systems.
This command is also used in dial-peer configuration mode to
affect an individual dial peer.
Step 5 exit (Optional) Exits voice-service configuration mode.
Example:
Router(conf-voi-serv)# exit
Step 6 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 1 voip
Example:
Router(config-dial-peer)# end
Restrictions
• Codecs on all the VoIP dial peers of the H.450 tandem gateway must be the same.
• Only one codec type is supported in the VoIP network at a time, and there are only two codec
choices: G.711 (A-law or mu-law) or G.729.
• Transcoding is not supported.
• Codec renegotiation is not supported. For example, if an H.323 call that uses a G.729 codec is
received by a Cisco Unified CME system and is forwarded to a voice-mail system that requires a
G.711 codec, the codec cannot be renegotiated from G.729 to G.711.
• H.323-to-SIP hairpin call routing is supported only with Cisco Unity Express. For more
information, see Integrating Cisco CallManager Express and Cisco Unity Express.
• Cisco Unified Communications Manager must use a media termination point (MTP), intercluster
trunk (ICT) mode, and slow start.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. allow-connections h323 to h323
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode to establish global call
transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 allow-connections h323 to h323 Enables VoIP-to-VoIP call connections. Use the no form of the
command to disable VoIP-to-VoIP connections; this is the default.
Example:
Router(conf-voi-serv)# allow-connections
h323 to h323
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-voi-serv)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. calling-number local
6. exit
7. voice service voip
8. allow connections from-type to to-type
9. supplementary-service h450.3
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# exit
Step 7 voice service voip Enters voice-service configuration mode.
Example:
Router(config)# voice service voip
Step 8 allow connections from-type to to-type Allows connections between specific types of endpoints in a
network.
Example: • from-type—Originating endpoint type. Valid choices are
Router(conf-voi-serv)# allow connections h323 and sip.
h323 to sip
• to-type—Terminating endpoint type. Valid choices are h323
and sip.
Step 9 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services capabilities
exchange globally. This is the default. Use the no form of this
command to disable H.450.3 capabilities globally. This command
Example:
Router(conf-voi-serv)# no
can also be used in dial-peer configuration mode to disable
supplementary-service h450.3 H.450.3 functionality for a single dial peer.
Note If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for
the dial peer.
Step 10 end Returns to privileged EXEC mode.
Example:
Router(config-voi-serv)# end
Prerequisites
Cisco Unified CME 4.0 or a later version.
Restrictions
• QSIG integration supports SCCP phones only.
• QSIG integration is exclusive; once QSIG integration is configured, QSIG transit node capability is
disabled. There is no dial-peer control to enable either transit or originate/terminate capability on a
call by call basis.
• If you enable QSIG supplementary services at a system-level, you cannot disable the capability on
individual dial peers.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. supplementary-service h450.7
5. qsig decode
6. exit
7. voice service pots
8. supplementary-service qsig call-forward
9. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-voi-serv)# qsig decode
Step 6 exit Exits VoIP voice-service configuration mode.
Example:
Router(config-voi-serv)# exit
Step 7 voice service pots Enters POTS voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service pots
Step 8 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
(ISO 13873) to forward calls to another number.
Example:
Router(config-voi-serv)# supplementary-service
qsig call-forward
Step 9 end Returns to privileged EXEC mode.
Example:
Router(config-voi-serv)# end
Prerequisites
Cisco Unified CME 4.0 or a later version.
Restrictions
• QSIG integration supports SCCP phones only.
• QSIG integration is exclusive; once QSIG integration is configured, QSIG transit node capability is
disabled. There is no dial-peer control to enable either transit or originate/terminate capability on a
call by call basis.
• If you enable QSIG supplementary services at a system-level, you cannot enable or disable the
capability on individual dial peers.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. qsig decode
5. exit
6. dial-peer voice tag voip
7. supplementary-service h450.7
8. exit
9. dial-peer voice tag pots
10. supplementary-service qsig call-forward
11. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters VoIP voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 qsig decode Enables decoding for QSIG supplementary services.
Example:
Router(config-voi-serv)# qsig decode
Step 5 exit Exits VoIP voice-service configuration mode.
Example:
Router(config-voi-serv)# exit
Example:
Router(config-dial-peer)# exit
Step 9 dial-peer voice tag pots Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 2 pots
Step 10 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
Router(config-dial-peer)# supplementary-service (ISO 13873) to forward calls to another number.
qsig call-forward
Step 11 end Returns to privileged EXEC mode.
Example:
Router(config-dial-peer)# end
Disabling SIP Supplementary Services for Call Forward and Call Transfer
To disable REFER messages for call transfers or redirect responses for call forwarding from being sent
to the destination by Cisco Unified CME, perform the following steps. You can disable these
supplementary features if the destination gateway does not support them.
Prerequisites
Cisco Unified CME 4.1 or a later version.
Restrictions
Disabling supplementary services is supported only when all endpoints are SCCP or all endpoints are
SIP. It does not support a mix of SCCP and SIP endpoints.
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode to set global
or parameters for VoIP features.
dial-peer voice tag voip or
Enters dial peer configuration mode to set parameters for a
Example: specific dial peer.
Router(config)# voice service voip
or
Router(config)# dial-peer voice 99 voip
Step 4 no supplementary-service sip {moved-temporarily Disables SIP call forwarding or call transfer supplementary
| refer} services globally or for a dial peer.
• moved-temporarily—SIP redirect response for call
Example: forwarding.
Router(conf-voi-serv)# no supplementary-service
sip refer • refer—SIP REFER message for call transfers.
or • Sending REFER and redirect messages to the
Router(config-dial-peer)# no destination is the default behavior.
supplementary-service sip refer
Note This command is supported for calls between SIP
phones and calls between SCCP phones. It is not
supported for a mixture of SCCP and SIP endpoints.
Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-voi-serv)# end
or
Router(config-dial-peer)# end
Figure 33 Network with Cisco Unified CME and Cisco Unified Communications Manager
IP IP
Cisco Unified CallManager 3
Phone 1 Phone 2
4001 4002 H.323 Connection
in ICT mode using slow start
PSTN
V V V
Telephone
IP IP IP IP IP IP
146621
Prerequisites
• Cisco Unified CME must be configured to forward calls using local hairpin routing. For
configuration information, see the “Forwarding Calls Using Local Hairpin Routing” section on
page 533.
Configuring Cisco CME 3.1 or Later to Interwork with Cisco Unified Communications Manager
All of the Cisco IOS commands in this section are optional because they are set by default to work with
Cisco Unified Communications Manager. They are included here only to explain how to implement
optional capabilities or return nondefault settings to their defaults.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. h323
5. telephony-service ccm-compatible
6. h225 h245-address on-connect
7. exit
8. supplementary-service h225-notify cid-update
9. exit
10. voice class h323 tag
11. telephony-service ccm-compatible
12. h225 h245-address on-connect
13. exit
14. dial-peer voice tag voip
15. supplementary-service h225-notify cid-update
16. voice-class h323 tag
17. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode to establish global
parameters.
Example:
Router(config)# voice service voip
Example:
Router(conf-voi-serv)# h323
Step 5 telephony-service ccm-compatible (Optional) Globally enables a Cisco CME 3.1 or later system to
detect a Cisco Unified Communications Manager and exchange
calls with it. This is the default.
Example:
Router(conf-serv-h323)# telephony-service • Use the no form of this command to disable
ccm-compatible Cisco Unified Communications Manager detection and
exchange. We do not recommend using the no form of the
command.
• Using this command in an H.323 voice class definition allows
you to specify this behavior for an individual dial peer.
Step 6 h225 h245-address on-connect (Optional) Globally enables a delay for the H.225 message
exchange of an H.245 transport address until a call is connected.
The delay allows the Cisco Unified Communications Manager to
Example:
Router(conf-serv-h323)# h225 h245-address
generate local ringback for calls to Cisco Unified CME phones.
on-connect This is the default.
• The no form of this command disables the delay. We do not
recommend using the no form of the command.
• Using this command in an H.323 voice class definition allows
you to specify this behavior for an individual dial peer.
Step 7 exit Exits H.323 voice-service configuration mode.
Example:
Router(conf-serv-h323)# exit
Step 8 supplementary-service h225-notify (Optional) Globally enables H.225 messages with caller-ID
cid-update updates to be sent to Cisco Unified Communications Manager.
This is the default.
Example: • The no form of the command disables caller-ID update. We
Router(conf-voi-serv)# do not recommend using the no form of the command.
supplementary-service h225-notify
cid-update This command is also used in dial-peer configuration mode to
affect a single dial peer.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for that dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for that dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for that
dial peer.
Step 9 exit Exits voice-service configuration mode.
Example:
Router(config-voice-service)# exit
Example:
Router(config-voice-class)# exit
Step 14 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode to set parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 28 voip
Step 15 supplementary-service h225-notify (Optional) Enables H.225 messages with caller-ID updates to
cid-update Cisco Unified Communications Manager for a specific dial peer.
This is the default.
Example: • The no form of the command disables caller-ID updates. We
Router(config-dial-peer)# no do not recommend using the no form of the command.
supplementary-service h225-notify
cid-update
Step 16 voice-class h323 tag (Optional) Applies the previously defined voice class with the
specified tag number to this dial peer.
Example:
Router(config-dial-peer)# voice-class
h323 48
Step 17 end Returns to privileged EXEC mode.
Example:
Router(config-dial-peer)# end
What to Do Next
Set up Cisco Unified Communications Manager using the configuration procedure in the “Enabling
Cisco Unified Communications Manager to Interwork with Cisco Unified CME” section on page 544.
Enabling Cisco Unified Communications Manager to Interwork with Cisco Unified CME
To enable Cisco Unified Communications Manager to interwork with a Cisco CME 3.1 or later system,
perform the following steps in addition to the normal Cisco Unified Communications Manager
configuration.
SUMMARY STEPS
DETAILED STEPS
Step 1 Set Cisco Unified Communications Manager service parameters. From Cisco Unified Communications
Manager Administration, choose Service Parameters. Choose the Cisco Unified Communications
Manager service, and make the following settings:
• Set the H323 FastStart Inbound service parameter to False.
• Set the Send H225 User Info Message service parameter to H225 Info for Ring Back.
Step 2 Configure the Cisco CME 3.1 or later system as an ICT in the Cisco Unified Communications Manager
network. For information about different intercluster trunk types and configuration instructions, see the
Cisco Unified Communications Manager documentation.
Step 3 Ensure that the Cisco Unified Communications Manager network uses an MTP. The MTP is required to
provide DSP resources for transcoding and for sending and receiving G.729 calls to the Cisco CME 3.1
or later system. All media streams between Cisco Unified Communications Manager and
Cisco CME 3.1 or later must pass through the MTP because Cisco CM 3.1 does not support transcoding.
For more information, see the Cisco Unified Communications Manager documentation.
Step 4 Set up dial peers to establish routing using the instructions in the Dial Peer Configuration on Voice
Gateway Routers guide.
Step 1 If you encounter lack of ringback on direct calls from a Cisco Unified Communications Manager phone
to an IP phone on a Cisco Unified CME system, check the show running-config command output to
ensure that the following two commands do not appear: no h225 h245-address on-connect and no
telephony-service ccm-compatible. These commands should be enabled, which is their default state.
Step 2 Use the debug h225 asn1 command to display the H.323 messages that are sent from the
Cisco Unified CME system to the Cisco Unified Communications Manager system to see if the H.245
address is being sent too early.
Step 3 For calls that are routed using VoIP-to-VoIP connections, use the show voip rtp connections detail
command to display the call identification number, IP addresses, and port numbers involved for all VoIP
call legs. This command includes VoIP-to-POTS and VoIP-to-VoIP call legs. The following is sample
output for this command:
Router# show voip rtp connections detail
Step 4 Use the show call prompt-mem-usage detail command to see information on ringback tone generation
that uses the interactive voice response (IVR) prompt playback mechanism. This ringback is needed for
hairpin transfers that are committed during the alerting-of-the-transfer-destination phase of the call and
for calls to destinations that do not provide in-band ringback tone, such as IP phones (FXS analog ports
do provide in-band ringback tone). Ringback tone is played to the transferred party by the
Cisco Unified CME system that performs the transfer (the system attached to the transferring party). The
system automatically generates tone prompts as needed based on the network-locale setting for the
Cisco Unified CME system.
If you are not getting ringback tone when you should, use the show call prompt-mem-usage command
to ensure that the correct prompt is loaded and playing. The following sample output indicates that a
prompt is playing (“Number of prompts playing”) and indicates the country code used for the prompt
(GB for Great Britain) and the codec.
Router# show call prompt-mem-usage detail
Prerequisites
• Connections between specific types of endpoints in a Cisco IP-to-IP gateway must be configured by
using the allow-connections command. For configuration information, see the “Enabling Calls in
Your VoIP Network” on page 96.
• Cisco CME 3.4 or a later version.
Restrictions
• SIP-to-SIP call forwarding is invoked only if that phone is dialed directly. Call forwarding is not
invoked when the phone number is called through a sequential, longest-idle, or peer hunt group.
• If call forwarding is configured for a hunt group member, call forward is ignored by the hunt group.
• In Cisco Unified CME 4.1 and later versions, Call Forward All requires SIP phones to be configured
with a directory number (using dn keyword in number command); direct line numbers are not
supported.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. call-forward b2bua all directory-number
5. call-forward b2bua busy directory-number
6. call-forward b2bua mailbox directory-number
7. call-forward b2bua noan directory-number timeout seconds
8. call-forward b2bua unreachable directory-number
9. end
DETAILED STEPS
Example:
Router# configure terminal
Prerequisites
• Cisco Unified CME 4.1 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.
• Call Forward All must be enabled on the directory number. For information, see “SIP: Configuring
SIP-to-SIP Phone Call Forwarding” on page 546.
Restrictions
• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
• If a user enables Call Forward All using the CfwdAll soft key, it is enabled on the primary line.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. call-feature-uri cfwdall service-uri
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Router(config)# voice register global
Step 4 call-feature-uri cfwdall service-uri Specifies the URI for soft keys on SIP phones connected to
a Cisco Unified CME router.
Example:
Router(config-register-global)#
call-feature-uri cfwdall
http://1.4.212.11/cfwdall
Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-register-global)# end
Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command must be enabled
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. phone-redirect-limit number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 phone-redirect-limit number Changes the default number of 3XX responses a SIP phone
that originates a call can handle for a single call.
Example: • Default: 5.
Router(config-register-global)#
phone-redirect-limit 8
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
Prerequisites
Cisco CME 3.4 or a later version.
Restrictions
• Blind transfer is not supported on Java-based phones, such as Cisco Unified IP Phone 7911G,
7941G, 7941GE, 7961G, 7961GE, 7970G, or 7971GE.
• In Cisco Unified CME 4.1, the soft key display can be customized for Java-based IP phones, such
as Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. For
configuration information, see “SCCP: Modifying Soft-Key Display” on page 832.
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to define
a template of common parameters for SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register template 1 • Range is 1 to 5.
Step 4 transfer-attended Enable a soft key for attended transfer on any supported SIP
phone that uses a template in which this command is
configure.
Example:
Router(config-register-template)#
transfer-attended
Step 5 transfer-blind Enable a soft key for blind transfer on any supported SIP
phone that uses a template in which this command is
configure.
Example:
Router(config-register-template)#
transfer-blind
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-template)# exit
Step 7 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example:
Router(config)# voice register pool 3
ephone-template 2
transfer max-length 8
ephone-dn 4
number 2977
ephone 6
button 1:4
ephone-template 2
H.450.12: Example
The following example globally disables H.450.12 capabilities and then enables them only on dial
peer 24.
voice service voip
no supplementary-service h450.12
!
dial-peer voice 24 voip
destination-pattern 555....
session target ipv4:10.5.6.7
supplementary-service h450.12
ephone-dn 25
number 74367
mwi qsig
call-forward all 74000
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
enable password pswd
!
aaa new-model
!
!
aaa session-id common
no ip subnet-zero
!
ip dhcp pool phone1
host 172.24.82.3 255.255.255.0
client-identifier 0100.07eb.4629.9e
default-router 172.24.82.2
option 150 ip 172.24.82.2
!
ip dhcp pool phone2
host 172.24.82.4 255.255.255.0
client-identifier 0100.0b5f.f932.58
default-router 172.24.82.2
option 150 ip 172.24.82.2
!
ip cef
no ip domain lookup
no mpls ldp logging neighbor-changes
no ftp-server write-enable
!
ephone-dn 1
number 3001
name abcde-1
call-forward busy 4001
!
ephone-dn 2
number 3002
name abcde-2
!
ephone-dn 3
number 3003
name abcde-3
!
ephone-dn 4
number 3004
name abcde-4
!
ephone 1
mac-address 0003.EB27.289E
button 1:1 2:2
!
ephone 2
mac-address 000D.39F9.3A58
button 1:3 2:4
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
password pswd
!
end
Building configuration...
Where to Go Next
If you are finished modifying the configuration, generate a new configuration file and restart the phones.
See “Generating Configuration Files for Phones” on page 245.
Soft Keys
To block the function of the call-forward-all or transfer soft key without removing the key display or to
remove the soft key from one or more phones, see the “How to Customize Soft Keys” section on
page 832.
Night Service
Calls can be automatically forwarded during night service hours, but you must define the night-service
periods, which are the dates or days and hours during which night service will be active. For instance,
you may want to designate night service periods that include every weeknight between 5 p.m. and 8 a.m.
and all day every Saturday and Sunday. For more information, see “Configuring Call-Coverage
Features” on page 563.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 27 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes features that can be used to provide appropriate, flexible coverage for incoming
calls in Cisco Unified Communications Manager Express (Cisco Unified CME).
Contents
• Information About Call Coverage Features, page 563
• How to Configure Call Coverage Features, page 585
• Configuration Examples for Call Coverage Features, page 619
• Where to Go Next, page 633
• Additional References, page 634
• Feature Information for Call Coverage Features, page 636
Call-Coverage Summary
Call coverage features are used to ensure that all incoming calls to Cisco Unified CME are answered by
someone, regardless of whether the called number is busy or does not answer.
Some single-dialed-number call-coverage features, such as hunt groups, can send incoming calls to a
single directory number to a pool of phone agents, while other features, such as call hunt, call waiting,
and call forwarding increase the chance of a call being answered by giving it another chance for a
connection if the dialed number is not available.
Multiple-dialed-number call-coverage features, such as call pickup, night service, and overlaid directory
numbers, provide different ways for one person to answer incoming calls to multiple numbers.
Any of the call-coverage features can be combined with other call-coverage features and with shared
lines and secondary numbers to design the call coverage plan that is best suited to your needs.
Table 28 summarizes call-coverage features.
Call Hunt
Call hunt allows you to use multiple directory numbers to provide coverage for a single called number.
You do this by assigning the same number to several primary or secondary ephone-dns or by using
wildcards in the number associated with the directory numbers.
Calls are routed based on a match between the number dialed and the destination patterns that are
associated with dial peers. Through the use of wildcards in destination patterns, multiple dial peers can
match a particular called number. Call hunt is the ability to search through the dial peers that match the
called number until the call is answered. Call hunt uses a technique called preference to control the order
in which dial peers are matched to an incoming call and a technique called huntstop to determine when
the search for another matching peer ends.
In Cisco Unified CME, incoming calls search through the virtual dial peers that are automatically
created when you define directory numbers. These virtual dial peers are not directly configurable; you
must configure the directory number to control call hunt for virtual dial peers.
Channel huntstop is used to stop the search for the two channels of a dual-line directory number. Channel
huntstop keeps incoming calls from hunting to the second channel if the first channel is busy or does not
answer. This keeps the second channel free for call transfer, call waiting, or three-way conferencing.
Huntstop prevents hunt-on-busy from redirecting a call from a busy phone into a dial peer that has been
setup with a catch-all default destination.
For configuration information, see the “SCCP: Configuring Call Hunt” section on page 586 and the
“SIP: Configuring Call Hunt” section on page 589.
Call Pickup
Call pickup, and pickup groups, enable phone users to answer a call that is ringing on a different
directory number other than their own. If both numbers to be answered are in the same pickup group, the
user presses fewer keys to pick up the call.
Call pickup has the following variations:
• Directed Call Pickup—Call pickup, explicit ringing extension. Any local phone user can pick up a
call that is on hold on another directory number in Cisco Unified CME. Phone user does not need to
belong to a pickup group to use this method. This is a default behavior.
• Group Pickup, Different Group—Call pickup, explicit group ringing extension. Phone user can
answer a ringing phone in any pickup group if the user knows the group number of the ringing
phone. If there is only one pickup group defined in Cisco Unified CME, the phone user can pick up
the call by pressing a soft key. Phone user does not need to belong to a pickup group to use this
method.
• Local Group Pickup—Call pickup, local group ringing extension. Phone users can pick up the called
number on another phone by pressing a soft key plus an asterisk (*) their own phone if both phones
are in the same pickup group.
Administrators can assign each ephone-dn independently to a maximum of one pickup group. There is
no limit to the number of ephone-dns that can be assigned to a single pickup group, and there is no limit
to the number of pickup groups that can be defined in a Cisco Unified CME system.
Pickup group numbers may be of varying length, but must have unique leading digits. For example, you
cannot define pickup group 17 and pickup group 177 for the same Cisco Unified CME system because
a pickup in group 17 will always be triggered before the user can enter the final 7 for 177.
ephone-dn 58
Call Pickup in the Same Group number 5558
.
1 Extension 5555 rings. 2 User at phone 2 presses GPickUp .
soft key and * (asterisk). .
ephone 1
Phone 1 Phone 2 mac-address 1111.1111.1111
Extension 5555 Extension 5556 button 1:55
IP IP
Pickup group 33 Pickup group 33
Phone 3 Phone 4 ephone 2
Extension 5557 Extension 5558 mac-address 2222.2222.2222
IP IP
Pickup group 44 No pickup group button 1:56
ephone 3
mac-address 3333.3333.3333
button 1:57
Call Pickup from a Different Group
ephone 4
mac-address 4444.4444.4444
1 Extension 5555 rings. 2 User at phone 3 presses
button 1:58
GPickUp soft key and dials 33.
.
.
Phone 1 Phone 2
.
Extension 5555 Extension 5556
IP IP
Pickup group 33 Pickup group 33
Phone 3 Phone 4
IP Extension 5557 IP Extension 5558
Pickup group 44 No pickup group
Phone 1 Phone 2
Extension 5555 Extension 5556
IP Pickup group 33 IP Pickup group 33
88954
This scenario assumes that every phone in the Cisco CME system is in pickup group
33, which differs slightly from the sample configuration shown to the right.
For configuration information, see the “SCCP: Creating Pickup Groups” section on page 591.
Call Waiting
Call waiting allows phone users to be alerted when they receive an incoming call while they are on
another call. Phone users hear a call-waiting tone when another party is trying to reach them and, on IP
phones, see the calling party information on the phone screen.
Call-waiting calls to IP phones with soft keys can be answered using the Answer soft key. Call-waiting
calls to analog phones controlled by Cisco Unified CME systems are answered using hookflash. When
phone users answer a call-waiting call, their original call is automatically put on hold. If a phone user
does not respond to a call-waiting notification, the call is forwarded as specified in the call-forward
noan command for that extension.
For an IP phone running SCCP, call waiting for single-line ephone-dns requires two ephone-dns to
handle the two calls. Call waiting on a dual-line ephone-dn requires only one ephone-dn because the two
channels of the ephone-dn handle the two calls. The audible call-waiting indicator can be either a
call-waiting beep or a call-waiting ring. For configuration information, see the “SCCP: Configuring
Call-Waiting Indicator Tone” section on page 593.
For a SIP phone, call waiting is automatically enabled when you configure a voice register pool. For SIP
phones directly connected to Cisco Unified CME, call waiting can be disabled at the phone-level. For
configuration information, see the “SIP: Enabling Call Waiting” section on page 595.
For information on call waiting using Overlaid ephone-dns, see the “Overlaid Ephone-dns” section on
page 581.
Incoming
Active Call Call Expected
Ephone-dn 1 Configuration Ephone-dn 2 Configuration on DN on DN Behavior
— no call-waiting beep DN 1 DN 2 No beep
no call-waiting beep — DN 1 DN 2 No beep
— no call-waiting beep generate DN 1 DN 2 No beep
— no call-waiting beep accept DN 1 DN 2 Beep
— no call-waiting beep accept DN 1 DN 2 No beep
no call-waiting beep generate
no call-waiting beep — DN 1 DN 1 No beep
no call-waiting beep generate — DN 1 DN 1 No beep
no call-waiting beep accept — DN 1 DN 1 No beep
no call-waiting beep accept no — DN 1 DN 1 No beep
call-waiting beep generate
no call-waiting beep generate — DN 1 DN 2 Beep
Incoming
Active Call Call Expected
Ephone-dn 1 Configuration Ephone-dn 2 Configuration on DN on DN Behavior
no call-waiting beep accept — DN 1 DN 2 No beep
— no call-waiting beep DN 1 DN 1 Beep
Hunt Groups
Hunt groups allow incoming calls to a specific number (pilot number) to be directed to a defined group
of directory numbers. Each hunt group can include up to 20 member directory numbers.
Incoming calls are redirected from a hunt group pilot number to the first directory number as defined by
the configuration. If the first directory number is busy or does not answer, the call is redirected to the
next phone in the list. A call continues to be redirected on busy or no answer from directory number to
directory number in the list until it is answered or until the call reaches the number that was defined as
the final number.
The redirect from one directory number to the next in the list is also known as a hop. You can set the
maximum number of redirects for specific peer or longest-idle hunt groups, and for the maximum
number of redirects allowed in a Cisco Unified CME system, both inside and outside hunt groups. If a
call makes the maximum number of hops or redirects without being answered, the call is dropped.
For information on displaying hunt group statistics, see Cisco Unified CME B-ACD and Tcl
Call-Handling Applications.
There are four different types of hunt groups. Each type uses a different strategy to determine the first
directory number that rings for successive calls to the hunt group pilot number. Hunt group types include
the following:
• Sequential Hunt Groups—Directory numbers always ring in the left-to-right order in which they are
listed when the hunt group is defined. The first number in the list is always the first number to be
tried when the pilot number is called. Maximum number of hops is not a configurable parameter for
sequential hunt groups.
• Peer Hunt Groups—The first directory number to ring is the number to the right of the directory number
that was the last to ring when the pilot number was last called. Ringing proceeds in a circular manner,
left to right, for the number of hops specified in the hunt group configuration.
• Longest-Idle Hunt Groups—Calls go first to the directory number that has been idle the longest for the
number of hops specified when the hunt group was defined. The longest-idle time is determined from
the last time that a phone registered, reregistered, or went on-hook.
• Parallel Hunt Groups—Calls ring all directory numbers in the hunt group simultaneously.
The number that is defined as the final number for a hunt group may also be the pilot number for another
hunt group (with suitable protection to avoid infinite loops). If a final number is assigned as the pilot
number of a second hunt group, the pilot number of the first hunt group cannot be configured as a final
number in any hunt group. If there is a third hunt group, the second hunt group cannot be configured as
a final number, and so forth.
Hunt-group chains can be configured in any length, but the actual number of hops that can be reached in
a chain is determined by the max-redirect command configuration. In the following example, a
maximum redirect number 15 or greater must be configured for callers to reach the final 5000 number.
If a lower number is configured, the call will disconnect.
ephone-hunt 1 sequential
pilot 8000
list 8001, 8002, 8003, 8004
final 9000
ephone-hunt 2 sequential
pilot 9000
list 9001, 9002, 9003, 9004
final 7000
ephone-hunt 3 sequential
pilot 7000
list 7001, 7002, 7003, 7004
final 5000
Figure 35 on page 571 illustrates a sequential hunt group, Figure 36 on page 572 illustrates a peer hunt
group, and Figure 37 on page 573 illustrates a longest-idle hunt group.
ephone-dn 88
1 Any phone dials the pilot number, 5601.
number 5001
2 Extension 5001, the leftmost number in the hunt group list, rings first ephone-dn 89
on phone 1. If extension 5001 is busy or does not answer, the call is number 5002
redirected to extension 5002 on phone 2.
3 If extension 5002 on phone 2 is busy or does not answer, the call is ephone-dn 90
redirected to extension 5017 on phone 3. number 5017
4 If phone 3 is busy or does not answer, the call is redirected to the final
number, extension 6000, which is associated with a voice-mail server. ephone 1
mac-address 1111.1111.1111
button 1:88
Any phone dials the pilot number.
IP ephone 2
mac-address 2222.2222.2222
6000 Voice-mail server button 1:89
5601
Pilot number
ephone 3
mac-address 3333.3333.3333
V button 1:90
88955
Phone 3
Button 1 is extension 5017
IP
1 Any phone dials the pilot number, 5601, which is not associated with a
physical phone instrument.
ephone-dn 88
2 Extension 5017 on phone 3 is selected to ring first because extension
5002 was the last number to ring the last time that the pilot number number 5001
was called.
ephone-dn 89
3 If extension 5017 is busy or does not answer, the call is redirected to number 5002
extension 5044 on phone 4 (first hop).
4 If extension 5044 is busy or does not answer, the call is redirected to ephone-dn 90
extension 5001 on phone 1 (second hop). number 5017
5 If extension 5001 is busy or does not answer, the call has reached the ephone-dn 91
maximum number of hops (3), and it is redirected to the final number, number 5044
extension 6000, which is associated with a voice-mail server.
ephone 1
Any phone dials the pilot number. mac-address 1111.1111.1111
IP button 1:88
Voice-mail server
Pilot number ephone 2
6000
5601 mac-address 2222.2222.2222
button 1:89
V ephone 3
mac-address 3333.3333.3333
Phone 1 button 1:90
Button 1 is extension 5001 IP
ephone 4
mac-address 4444.4444.4444
Phone 2
Button 1 is extension 5002 IP button 1:91
ephone-hunt 1 peer
Phone 3 pilot 5601
Button 1 is extension 5017
IP
list 5001, 5002, 5017, 5044
final 6000
Phone 4 hops 3
Button 1 is extension 5044 IP preference 1
88956
timeout 30
no-reg
1 Any phone dials the pilot number, 5601, which is not associated with a
physical phone instrument. ephone-dn 88
number 5001
2 Extension 5001 on phone 1 is selected to ring first because it has
been idle the longest. ephone-dn 89
3 If extension 5001 does not answer, the call is redirected to extension number 5002
5002 on phone 2 because it has been idle the longest (first hop).
ephone-dn 90
4 If extension 5002 does not answer, the call is redirected to extension
number 5017
5044 on phone 4 because it has been idle the longest (second hop).
5 If extension 5044 does not answer, the call has reached the maximum ephone-dn 91
number of hops (3), and it is redirected to the final number, extension 6000, number 5044
which is associated with a voice-mail server
ephone 1
Any phone dials the pilot number. mac-address 1111.1111.1111
IP button 1:88
Voice-mail server
Pilot number ephone 2
6000
5601 mac-address 2222.2222.2222
button 1:89
V ephone 3
mac-address 3333.3333.3333
Phone 1 button 1:90
Button 1 is extension 5001 IP
ephone 4
mac-address 4444.4444.4444
Phone 2
Button 1 is extension 5002 IP button 1:91
ephone-hunt 1 longest-idle
Phone 3 pilot 5601
Button 1 is extension 5017
IP
list 5001, 5002, 5017, 504
final 6000
Phone 4 hops 3
Button 1 is extension 5044 IP preference 1
103299
timeout 30
no-reg
The number of ringing calls that a parallel hunt group can support depends on whether call-waiting is
enabled on the SIP phones.
If call-waiting is enabled (the default), parallel hunt groups support multiple calls up to the limit of
call-waiting calls supported by a particular SIP phone model. You may not want to use unlimited
call-waiting however with parallel hunt-groups if agents do not want a large number of waiting calls
when they are already handling a call.
If call waiting is disabled, parallel hunt groups support only one call at a time in the ringing state. After
a call is answered (by one of the phones in the hunt group), a second call is allowed. The second and
subsequent calls ring only the idle phones in the hunt group, and bypass the busy phone that answered
the first call (because this phone is connected to the first call). After the second call is answered, a third
call is allowed, and so on until all the phones in the parallel hunt group are busy. The hunt group does
not accept further calls until at least one phone returns to the idle/on-hook state.
When two or more phones within the same parallel hunt group attempt to answer the same call, only one
phone can connect to the call. Phones that fail to connect must return to the on-hook state before they
can receive subsequent calls. Calls that arrive before a phone is placed on-hook are not presented to the
phone. For example, if a second call arrives after Phone 1 has answered the original call, but before
Phone 2 goes back on-hook, the second call bypasses Phone 2 (because it is offhook).
When a phone returns to the idle/on-hook state, it does not automatically re-synchronize to the next call
waiting to be answered. For example, in the previous scenario, if the second call is still ringing Phone 3
when Phone 2 goes on-hook, Phone 2 does not ring because it was offhook when the second call arrived.
For configuration information, see the “SIP: Configuring Hunt Groups” section on page 605.
Comparison
Factor Dynamic Membership Agent Status Control Automatic Agent Status Not-Ready
Purpose Allows an authorized agent to join Allows an agent to manually Automatically puts an agent’s
and leave hunt groups. activate a toggle to temporarily phone in a not-ready state after a
enter a not-ready state, in which specified number of hunt-group
hunt-group calls bypass the calls are unanswered by the
agent’s phone. agent’s phone.
Example Agent A joins a hunt group at Agent A takes a coffee break at Agent B is suddenly called away
8 a.m. and takes calls until 1 p.m., 10 a.m. and puts his phone into a from her desk before she can
when he leaves the hunt group. not-ready status while he is on manually put her phone into the
While Agent A is a member of the break. When he returns he puts his not-ready status. After a
hunt group, he occupies one of the phone back into the ready status hunt-group call is unanswered at
wildcard slots in the list of and immediately starts receiving Agent B’s phone, the phone is
numbers configured for the hunt hunt-group calls again. He automatically placed in the
group. At 1 p.m., Agent B joins retained his wildcard slot while he not-ready status and it is not
the hunt group using the same was in the not-ready status. presented with further hunt-group
wildcard slot that Agent A calls. When Agent B returns, she
relinquished when he left. manually puts her phone back into
the ready status.
Hunt-group slot An agent joining a hunt group An agent who enters the not-ready An agent who enters the not-ready
availability occupies a wildcard slot in the state does not give up a slot in the does not give up a slot in the hunt
hunt group list. An agent leaving hunt group. The agent continues to group. The agent continues to
the group relinquishes the slot, occupy the slot regardless of occupy the slot regardless of
which becomes available for whether the agent is in the whether the agent is in the
another agent. not-ready status. not-ready status.
Comparison
Factor Dynamic Membership Agent Status Control Automatic Agent Status Not-Ready
Agent activation An authorized agent uses a feature An agent uses the HLog soft key An agent who is a member of a
method access code (FAC) to join a hunt to toggle agent status between hunt group configured with the
group and a different FAC to leave ready and not ready. Agents can auto logout command does not
the hunt group. also use the HLog ephone FAC or answer the specified number of
the HLog ephone-dn FAC to calls, and the agent’s phone is
toggle between ready and automatically changed to the
not-ready if FACs are enabled. not-ready status. The agent uses
the HLog soft key or a FAC to
If the HLog soft key is not
return to the ready status.
enabled, the DND soft key can be
used to put an agent in the If the HLog soft key or FAC has
not-ready status and the agent will not been enabled in the
not receive any calls. configuration, the agent uses the
DND soft key to return to the
ready status.
Configuration The system administrator uses the The system administrator uses the The system administrator uses the
list command to configure up to HLog keyword with the auto logout command to enable
20 wildcard slots in a hunt group hunt-group logout command to automatic agent status not-ready
and uses the ephone-hunt login provide an HLog soft key on for a hunt group.
command to authorize certain display phones and uses the fac
This functionality is disabled by
directory numbers to use these command to enable standard FACs
default.
wildcard slots. or create a custom FAC.
See SCCP: Configuring Hunt
See SCCP: Configuring Hunt See SCCP: Configuring Hunt
Groups, page 596.
Groups, page 596. Groups, page 596.
Optional The system administrator can The system administrator can use The system administrator can use
customizations establish custom FACs for agents the softkeys commands to change the auto logout command to
to use to enter or leave a hunt the position or prevent the display specify the number of unanswered
group. of the HLog soft key on individual calls that will trigger an agent
phones. status change to not-ready and
whether this feature applies to
dynamic hunt-group members,
static hunt-group members, or
both.
The system administrator can use
the hunt-group logout command
to specify whether an automatic
change to the not-ready status also
places a phone in DND mode.
ephone-hunt 25 sequential
pilot 7000
list 7001, 7002, *, *
description Service Group
final 9000
To leave a hunt group, a phone user dials the standard or custom FAC for leaving a hunt group. The
standard FAC to leave a hunt group is #3. See “Customizing Soft Keys” on page 829.
Note The Dynamic Membership feature is different from the Agent Status Control feature and the Automatic
Agent Status Not-Ready feature. Table 30 on page 575 compares the features.
Agents use the HLog soft key or the DND soft key to put a phone into the not-ready status. When the
HLog soft key is used to put a phone in the not-ready status, it does not receive hunt group calls but can
receive other calls. If the DND soft key is used, the phone does not receive any calls until it is returned
to the ready status. The HLog and DND soft keys toggle the feature: if the phone is in the ready status,
pressing the key puts the phone in the not-ready status and vice-versa.
The DND soft key is visible on phones by default, but the HLog soft key must be enabled in the
configuration using the hunt-group logout command, which has the following options:
• HLog—Enables both an HLog soft key and a DND soft key on phones in the idle, seized, and
connected call states. When you press the HLog soft key, the phone is changed from the ready to
not-ready status or from the not-ready to ready status. When the phone is in the not-ready status, it
does not receive calls from the hunt group, but it is still able to receive calls that do not come through
the hunt group (calls that directly dial its extension). The DND soft key is also available to block all
calls to the phone if that is the preferred behavior.
• DND—Enables only a DND soft key on phones. The DND soft key also changes a phone from the
ready to not-ready status or from the not-ready to ready status, but the phone does not receive any
incoming calls, including those from outside hunt groups.
Phones without soft-key displays can use a FAC to toggle their status from ready to not-ready and back
to ready. The fac command must be used to enable the standard set of FACs or to create custom FACs.
The standard FAC to toggle the not-ready status at the directory number (extension) level is *4 and the
standard FAC to toggle the not-ready status at the ephone level (all directory numbers on the phone) is
*5. See Where to Go Next, page 633.
Note The Agent Status Control feature is different from the Dynamic Membership feature and the Automatic
Agent Status Not-Ready feature. Table 30 on page 575 compares the features.
Note The Automatic Agent Status Not-Ready feature is different from the Dynamic Membership feature and
the Agent Status Control feature. Table 30 on page 575 compares the features.
Night Service
The night-service feature allows you to provide coverage for unstaffed extensions during hours that you
designate as “night-service” hours. During the night-service hours, calls to the designated extensions
(known as night-service directory numbers or night-service lines) send a special “burst” ring to phones
that have been specified to receive this special ring (the phones are known as night-service phones).
Phone users at the night-service phones can then use the call-pickup feature to answer the incoming calls
from the night-service directory numbers (Figure 38).
For example, the night-service feature can allow an employee working after hours to intercept and
answer calls that are presented to an unattended receptionist’s phone. This feature is useful for sites at
which all incoming public switched telephone network (PSTN) calls have to be transferred by a
receptionist because the PSTN connection to the Cisco Unified CME system does not support Direct
Inward Dialing (DID). When a call arrives at the unattended receptionist’s phone during hours that are
specified as night service, a ring burst notifies a specified set of phones of the incoming call. A phone
user at any of the night-service phones can intercept the call using the call-pickup feature. Night-service
call notification is sent every 12 seconds until the call is either answered or aborted.
If optionally configured, night service can be manually toggled on and off from any phone that has a line
that is designated as a night-service line. When night service is active, a message is displayed on the
night-service phones.
Night service requires that you define the following parameters:
1. Night-service time period—Day or date and hours during which night service is active. Step 4 to
Step 8 in the following procedure define the night-service period.
2. Night-service extensions (directory numbers)—When a night-service extension receives an
incoming call during the night-service period, night-service notification is triggered. Step 12 in the
following procedure specifies night service for an directory number.
3. Night-service notification phones (ephones)—Night-service notification phones are alerted with a
distinctive ring when incoming calls are received on night-service lines during the night-service
time period. The night-service notification phone user can answer the call using call pickup or group
call pickup. Step 15 in the following procedure assigns night-service notification to a phone. This
phone receives a distinctive alerting ring and notification display when a night-service extension
receives an incoming call.
4. (Optional) Night-service toggle code—A code to allow night-service treatment to be manually
toggled off and on from any phone that has a line assigned to night service. Before Cisco CME 3.3,
using the night-service code turned night service on or off only for directory numbers on the phone
at which the code was entered. In Cisco CME 3.3 and later versions, using the night-service code at
any phone with a night-service directory number turns night service on or off for all phones with
night-service directory numbers. The following procedure defines a night-service toggle code.
Phone 5
1 Extension 1000 has been designated as a night-service Button 1 is extension 1000
IP
extension (ephone-dn). When extension 1000 receives an Extension 1000 is a night-
incoming call during a night-service period, phone 5 rings service extension
and notification is made to the night-service phones.
telephony-service IP
night-service day fri 17:01 17:00
Phone 14
night-service day sat 17:01 17:00
Button 1 is extension 1010
night-service day sun 17:01 07:59
Phone 14 is a night-service phone
night-service date jan 1 00:00 00:00
night-service code *1234
!
ephone-dn 1 IP
number 1000
night-service bell Phone 15
! Button 1 is extension 1011
ephone-dn 10 Phone 15 is a night-service phone
number 1010
!
ephone-dn 11
number 1011
!
ephone 5
mac-address 1111.2222.0001
button 1:1
!
ephone 14
mac-address 1111.2222.0002
button 1:10
night-service bell
!
ephone 15
mac-address 1111.2222.0003
88951
button 1:11
night-service bell
Overlaid Ephone-dns
Overlaid ephone-dns are directory numbers that share the same button on a phone. Overlaid ephone-dns
can be used to receive incoming calls and place outgoing calls. Up to 25 ephone-dns can be assigned to
a single phone button. They can have the same extension number or different numbers. The same
ephone-dns can appear on more than one phone and more than one phone can have the same set of
overlaid ephone-dns.
The order in which overlaid ephone-dns are used by incoming calls can be determined by the call hunt
commands, preference and huntstop. For example, ephone-dn 1 to ephone-dn 4 have the same
extension number, 1001. Three phones are configured with the button 1o1,2,3,4 command. A call to
1001 will ring on the ephone-dn with the highest preference and display the caller ID on all phones that
are on hook. If another incoming call to 1001 is placed while the first call is active (and the first
ephone-dn with the highest preference is configured with the no huntstop command), the second call
will roll over to the ephone-dn with the next-highest preference, and so forth. For more information, see
the “Call Hunt” section on page 565.
If the ephone-dns in an ephone-dn overlay use different numbers, incoming calls go to the ephone-dn
with the highest preference. If no preferences are configured, the dial-peer hunt command setting is
used to determine which ephone-dns are used for incoming calls. The default setting for the dial-peer
hunt command is to randomly select an ephone-dn that matches the called number.
Note To continue or to stop the search for ephone-dns, you must use, respectively, the no huntstop and
huntstop commands under the individual ephone-dns. The huntstop setting is applied only to the dial
peers affected by the ephone-dn command in telephony-service mode. Dial peers configured in global
configuration mode comply with the global configuration huntstop setting.
Figure 39 on page 581 shows an overlay set with two directory numbers and one number that is shared
on two phones. Ephone-dn 17 has a default preference value of 0, so it will receive the first call to
extension 1001. The phone user at phone 9 answers the call, and a second incoming call to
extension 1001 can be answered on phone 10 using directory number 18.
Phone 9 ephone-dn 17
Button 1 is two appearances number 1001
of extension 1001
ephone-dn 18
IP number 1001
preference 1
IP V ephone 9
Phone 10 button 1o17,18
88894
When a call is answered on an ephone-dn, that ephone-dn is no longer available to other phones that
share the ephone-dn in overlay mode. For example, if extension 1001 is answered by phone 1, caller ID
for extension 1001 displays on phone 1 and is removed from the screens of phone 2 and phone 3. All
actions pertaining to the call to extension 1001 (ephone-dn 17) are displayed on phone 1 only. If phone
1 puts extension 1001 on hold, the other phones will not be able to pick up the on-hold call using a simple
shared-line pickup. In addition, none of the other four phones will be able to make outgoing calls from
the ephone-dn while it is in use. When phone users press button 1, they will be connected to the next
available ephone-dn listed in the button command. For example, if phone 1 and phone 2 are using
ephone-dn 1 and ephone-dn 2, respectively, phone 3 must pick up ephone-dn 3 for an outgoing call.
If there are more phones than ephone-dns associated with an ephone-dn overlay set, it is possible for
some phones to find that all the ephone-dns within their overlay set are in use by other phones. For
example, if five phones have a line button configured with the button 1o1, 2, 3 command, there may be
times when all three of the ephone-dns in the overlay set are in use. When that occurs, the other two
phones will not be able to use an ephone-dn in the overlay set. When all ephone-dns in an overlay set
are in use, phones with this overlay set will display the remote-line-in-use icon (a picture of a phone with
a flashing X through it) for the corresponding line button. When at least one ephone-dn becomes
available within the overlay set (that is, an ephone-dn is either idle or ringing), the phone display reverts
to showing the status of the available ephone-dn (idle or ringing).
Shared-Line Overlays
Dual-line ephone-dns can also use overlays. The configuration parameters are the same as for single-line
ephone-dns, except that the huntstop channel command must be used to keep calls from hunting to the
ephone-dn’s second channel.
The primary ephone-dn in a shared-line overlay set should be unique to the phone to guarantee that the
phone has a line available for outgoing calls, and to ensure that the phone user can obtain dial-tone even
when there are no idle lines available in the rest of the shared-line overlay set. Use a unique ephone-dn
to provide for a unique calling party identity on outbound calls made by the phone so that the called user
can see which specific phone is calling.
The following example shows the configuration for a simple shared-line overlay set. The primary
ephone-dn that is configured for each phone is unique while the remaining ephone-dns 10, 11, and 12
are shared in the overlay set on both phones:
!
ephone 1
mac-address 1111.1111.1111
button 1o1,10,11,12
!
ephone 2
mac-address 2222.2222.2222
button 1o2,10,11,12
For a more detailed example, see the “Shared-line Overlaid Ephone-dns: Example” section on page 628.
A more complex directory number configuration mixes overlaid directory numbers with shared directory
numbers and plain dual-line directory numbers on the same phones. Figure 40 on page 583 illustrates
the following example of a manager with two assistants. On the manager’s phone the same number,
2001, appears on button 1 and button 2. The two line appearances of extension 2001 use two single-line
directory numbers, so the manager can have two active calls on this number simultaneously, one on each
button. The directory numbers are set up so that button 1 will ring first, and if a second call comes in,
button 2 will ring. Each assistant has a personal directory number and also shares the manager’s
directory numbers. Assistant 1 has all three directory numbers in an overlay set on one button, whereas
assistant 2 has one button for the private line and a second button with both of the manager’s lines in an
overlay set. A sequence of calls might be as follows.
1. An incoming call is answered by the manager on extension 2001 on button 1 (directory number 20).
2. A second call rings on 2001 and rolls over to the second button on the manager’s phone (directory
number 21). It also rings on both assistants’ phones, where it is also directory number 21, a shared
directory number.
3. Assistant 2 answers the call. This is a shared overlay line (one directory number, 21, is shared among
three phones, and on two of them this directory number is part of an overlay set). Because it is shared
with button 2 on the manager’s phone, the manager can see when assistant 2 answers the call.
4. Assistant 1 makes an outgoing call on directory number 22. The button is available because of the
additional directory numbers in the overlay set on the assistant 1 phone.
At this point, the manager is in conversation on directory number 20, assistant 1 is in conversation on
directory number 22, and assistant 2 is in conversation on directory number 21.
Manager phone
Button 1 is extension 2001 ephone-dn 20
Button 2 is extension 2001 number 2001
no huntstop
IP ! Manager number
ephone-dn 21
number 2001
IP V preference 1
Assistant 1 phone ! Manager number
Button 1 is extension 2001
and extension 2002 ephone-dn 22
number 2002
! Assistant 1 personal number
IP
Assistant 2 phone ephone-dn 23
Button 1 is extension 2003 number 2003
Button 2 is extension 2001 ! Assistant 2 personal number
ephone 8
button 1:20 2:21
! Manager phone
ephone 9
button 1o22,20,21
! Assistant 1 phone
ephone 10
88895
button 1:23 2o20,21
! Assistant 2 phone
For configuration information, see the “SCCP: Configuring Overlaid Ephone-dns” section on page 615.
The behavior of overlaid ephone-dns with call waiting and overlaid ephone-dns without call waiting is
the same, except for the following:
• Calls to numbers included in overlaid ephone-dns with call waiting will cause inactive phones to
ring and active phones connected to other parties to generate auditory call-waiting notification. The
default sound is beeping, but you can configure an ephone-dn to use a ringing sound. (See the
“SCCP: Configuring Call-Waiting Indicator Tone” section on page 593.) Visual call-waiting
notification includes the blinking of handset indicator lights and the display of caller IDs.
For example, if three of four phones are engaged in calls to numbers from the same overlaid
ephone-dn with call-waiting and another call comes in, the one inactive phone will ring, and the
three active phones will issue auditory and visual call-waiting notification.
• In Cisco Unified CME 4.0 and later versions, up to six waiting calls can be displayed on
Cisco Unified IP Phone 7940G, 7941G, 7941G-GE, 7960G, 7961G, 7961G-GE, 7970G, and
7971G-GE. For all other phones and earlier Cisco Unified CME versions, two calls to numbers in
an overlaid ephone-dn set can be announced. Subsequent calls must wait in line until one of the two
original calls has ended. The callers who are waiting in the line will hear a ringback tone.
For example, a Cisco Unified IP Phone 7910 (maximum two call-waiting calls) has a button configured
with a set of overlaid ephone-dns with call waiting (button 1c1,2,3,4). A call to ephone-dn 1 is
answered. A call to ephone-dn 2 generates call-waiting notification. Calls to ephone-dn 3 and ephone-dn
4 will wait in line and remain invisible to the phone user until one of the two original calls ends. When
the call to ephone-dn 1 ends, the phone user can then talk to the person who called ephone-dn 2. The call
to ephone-dn 3 issues call-waiting notification while the call to ephone-dn 4 waits in line. (The
Cisco Unified IP Phone 7960 supports six calls waiting.) Phones configured for call waiting do not
generate call-waiting notification when they are transferring calls or hosting conference calls.
Note that if an overlaid ephone-dn has call-forward-no-answer configured, calls to the ephone-dn that
are unanswered before the no-answer timeout expires are forwarded to the configured destination. If
call-forward-no-answer is not configured, incoming calls receive ringback tones until the calls are
answered.
More than one phone can use the same set of overlaid ephone-dns. In this case, the call-waiting behavior
is slightly different. The following example demonstrates call waiting for overlaid ephone-dns that are
shared on two phones.
ephone 1
button 1c1,2,3,4
!
ephone 2
button 1c1,2,3,4
1. A call to ephone-dn 1 rings on ephone 1 and on ephone 2. Ephone 1 answers, and the call is no longer
visible to ephone 2.
2. A call to ephone-dn 2 issues a call-waiting notification to ephone 1 and rings on ephone 2, which
answers. The second call is no longer visible to ephone 1.
3. A call to ephone-dn 3 issues a call-waiting notification to ephone 1 and ephone 2. Ephone 1 puts the
call to ephone-dn 1 on hold and answers the call to ephone-dn 3. The call to ephone-dn 3 is no longer
visible to ephone 2.
4. A call to ephone-dn 4 is issues a call-waiting notification on ephone 2. The call is not visible on
ephone 1 because it has met the two-call maximum by handling the calls to ephone-dn 1 and
ephone-dn 3. (Note that the call maximum is six for those phones that are able to handle six
call-waiting calls, as previously described.)
Note Ephone-dns accept call interruptions, such as call waiting, by default. For call waiting to work, the
default must be active. For more information, see the “SCCP: Configuring Call-Waiting Indicator Tone”
section on page 593.
Extending Calls for Overlaid Ephone-dns to Other Buttons on the Same Phone
Phones with overlaid ephone-dns can use the button command with the x keyword to dedicate one or
more additional buttons to receive overflow calls. If an overlay button is busy, an incoming call to any
of the other ephone-dns in the overlay set rings on the first available overflow button on each phone that
is configured to receive the overflow. This feature works only for overlaid ephone-dns that are
configured with the button command and the o keyword; it is not supported with overlaid ephone-dns
that are configured using the button command and the c keyword or other types of ephone-dns that are
not overlaid.
Using the button command with the c keyword results in multiple calls on one button (the button is
overlaid with multiple ephone-dns that have call waiting), whereas using the button command with the
o keyword and the x keyword results in one call per button and calls on multiple buttons.
For example, an ephone has an overlay button with ten numbers assigned to it using the button command
and the o keyword. The next two buttons on the phone are configured using the button command and
the x keyword. These buttons are reserved to receive additional calls to the overlaid extensions on the
first button when the first button is in use.
ephone 276
button 1o24,25,26,27,28,29,30,31,32,33 2x1 3x1
For configuration information, see the “SCCP: Configuring Overlaid Ephone-dns” section on page 615.
Call Hunt
• SCCP: Configuring Call Hunt, page 586 (required)
• SCCP: Verifying Call Hunt, page 587 (optional)
• SIP: Configuring Call Hunt, page 589 (required)
Call Pickup
• SCCP: Creating Pickup Groups, page 591 (required)
• SCCP: Verifying Call Pickup, page 592 (optional)
Call Waiting
• SCCP: Configuring Call-Waiting Indicator Tone, page 593 (optional)
• SIP: Enabling Call Waiting, page 595 (required)
Hunt Groups
• SCCP: Configuring Hunt Groups, page 596 (required)
• SCCP: Verifying Hunt Groups, page 603 (optional)
Night Service
• SCCP: Configuring Night Service, page 609 (required)
• SCCP: Verifying Night Service, page 613 (optional)
Overlaid Ephone-dns
• SCCP: Configuring Overlaid Ephone-dns, page 615 (required)
• SCCP: Verifying Overlaid Ephone-dns, page 618 (optional)
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. preference preference-order [secondary secondary-order]
6. huntstop
7. huntstop channel
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode for the purpose of
configuring a directory number.
Example:
Router(config)# ephone-dn 20 dual-line
or
Router(config-ephone-dn)# huntstop
Step 7 huntstop channel (Optional) Enables channel huntstop, which keeps a call from
hunting to the next channel of an ephone-dn if the first channel is
busy or does not answer.
Example:
Router(config-ephone-dn)# huntstop • Required for dual-line ephone-dns.
channel
Step 8 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
What to Do Next
If you want to collect statistics for hunt groups, see Cisco Unified CME B-ACD and Tcl Call-Handling
Applications.
SUMMARY STEPS
1. show running-config
2. show telephony-service ephone-dn
DETAILED STEPS
ephone-dn 2 dual-line
number 126
description FrontDesk
name Receptionist
preference 1
call-forward busy 500
huntstop channel
no huntstop
ephone-dn 243
number 1233
preference 1
huntstop
!
dial-peer voice 20026 pots
destination-pattern 5002
huntstop
call-forward noan 5001 timeout 45
port 50/0/2
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. preference preference-order
6. huntstop
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config)# voice register dn 1
Step 4 number number Associates a phone number with the directory number.
• Assign the same number to several directory numbers
Example: to create a group of virtual dial peers through which the
Router(config-register-dn)# number 5001 incoming called number must search.
Step 5 preference preference-order Creates the preference order for matching the VoIP dial
peers created for the number associated with this directory
number to establish the hunt strategy for incoming calls.
Example:
Router(config-register-dn)# preference 4 • Default is 0, which is the highest preference.
What to Do Next
If you want to collect statistics for hunt groups, see Cisco Unified CME B-ACD and Tcl Call-Handling
Applications.
Note To selectively disable directed call pickup for one or more SCCP phones, use the features blocked
command in ephone-template mode. For configuration information, see “SCCP: Enabling Ephone
Templates” on page 883.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. no service directed-pickup
5. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# telephony-service
Step 4 no service directed-pickup Disables directed call pickup.
• Changes the action of the PickUp soft key to perform
Example: local group call pickup rather than directed call pickup.
Router(config-telephony)# no service
directed-pickup
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Prerequisites
Directory numbers to be added to a pickup group must be configured in Cisco Unified CME. For
configuration information, see “SCCP: Creating Directory Numbers” on page 158.
Restrictions
• Each directory number can be independently assigned to a maximum of one pickup group.
• There is no limit to the number of directory numbers that can be assigned to a single pickup group.
• There is no limit to the number of pickup groups that can be defined in Cisco Unified CME.
• Pickup group numbers may be of varying length, but must have unique leading digits.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. pickup-group number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode for the purpose of
configuring a directory number.
Example:
Router(config)# ephone-dn 20 dual-line
Step 4 pickup-group number Creates a pickup group and assigns the directory number
being configured to the group.
Example: • number—Digit string of up to 32 characters. Group
Router(config-ephone-dn)# pickup-group 2345 numbers may be of varying length, but they must have
unique leading digits. For example, if there is a group
number 17, there cannot also be a group number 177.
Step 5 exit Exits ephone-dn configuration mode.
Example:
Router(config-ephone-dn)# exit
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Step 2 Use the show telephony-service ephone-dn command to display call pickup configuration information.
Router# show telephony-service ephone-dn
ephone-dn 2
number 5002
pickup group 30
call-forward noan 5001 timeout 8
Restrictions
• The call-waiting ring option is not supported if the ephone-dn is configured with the no call-waiting
beep accept command.
• If you configure a button to have a silent ring, you will not hear a call-waiting beep or call-waiting
ring regardless of whether the ephone-dn associated with the button is configured to generate a
call-waiting beep or call-waiting ring. To configure a button for silent ring, see the “SCCP:
Assigning Directory Numbers to Phones” on page 160.
• The call-waiting beep volume cannot be adjusted through Cisco Unified CME for the
Cisco Unified IP Phone 7902G, Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G,
Cisco ATA-186, and Cisco ATA-188.
• The call-waiting ring option is not supported on the Cisco Unified IP Phone 7902G,
Cisco Unified IP Phone 7905G, or Cisco Unified IP Phone 7912G.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. call-waiting beep [accept | generate]
5. call-waiting ring
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example:
Router(config)# ephone-dn 20 dual-line
Step 4 call-waiting beep [accept | generate] Enables an ephone-dn to generate or accept call-waiting
beeps.
Example: • Default is directory number both accepts and generates
Router(config-ephone-dn)# no call-waiting beep call waiting beep.
accept
• The beep is heard only if the other ephone-dn is
configured to accept call-waiting beeps (default).
Step 5 call-waiting ring (Optional) Enables an ephone-dn to use a ring indicator for
call-waiting notification.
Example: • To use this command, do not disable call-waiting beep
Router(config-ephone-dn)# call-waiting ring by using the no call-waiting beep accept command.
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
Step 2 Use the show telephony-service ephone-dn command to display call-waiting configuration
information.
Router# show telephony-service ephone-dn
ephone-dn 1 dual-line
number 126 secondary 1261
preference 0 secondary 9
no huntstop
huntstop channel
call-forward busy 500 secondary
call-forward noan 500 timeout 10
call-waiting beep
Prerequisites
• Cisco Unified CME 3.4 or a later version.
• mode cme command must be configured in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. call-waiting
5. exit
6. voice register global
7. hold-alert timeout
8. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-register-pool)# exit
Step 6 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 7 hold-alert timeout Sets an audible alert notification when a call is on hold on a
SIP phone. Default is disabled.
Example: • timeout—Interval after which an audible alert
Router(config-register-global)# hold-alert 30 notification is repeated, in seconds. Range: 15 to 300.
Step 8 end Exits to privileged EXEC mode.
Example:
Router(config-register-global)# end
Prerequisites
Directory numbers to be included in a hunt group must be already configured in Cisco Unified CME.
For configuration information, see “SCCP: Creating Directory Numbers” on page 158.
Restrictions
• The HLog soft key is available only on display phones. It is not available on Cisco Unified IP
Phones 7902, 7905, and 7912; Cisco IP Communicator; and Cisco VG 224.
• Shared ephone-dns cannot use the Agent Status Control or Automatic Agent Not-Ready feature.
• The Agent Status Control feature and the HLog soft key require the user locale to be set to US. To
display the HLog soft key on a Cisco Unified IP Phone 7940 or Cisco Unified IP Phone 7960,
change the user locale to any locale other than US and reset the phone. Then change the user locale
to US and reset the phone again.
• If directory numbers that are members of a hunt group are to be configured for called-name display,
the following restrictions apply:
– The primary or secondary pilot number must be defined using at least one wildcard character.
– The phone numbers in the list command cannot contain wildcard characters.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-hunt hunt-tag {longest-idle | peer | sequential}
4. pilot number [secondary number]
5. list number[, number...]
6. final final-number
7. hops number
8. timeout seconds[, seconds...]
9. max-timeout seconds
10. preference preference-order [secondary secondary-order]
11. no-reg [both | pilot]
12. fwd-final {orig-phone | final}
13. forward local-calls
14. secondary start [current | next | agent-position]
15. present-call {idle-phone | onhook-phone}
16. from-ring
17. description text-string
18. display-logout text-string
19. exit
20. telephony-service
21. max-redirect number
22. hunt-group logout {DND | HLog}
23. exit
24. ephone-dn dn-tag
25. ephone-hunt login
26. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-hunt hunt-tag {longest-idle | peer | Enters ephone-hunt configuration mode to define an ephone
sequential} hunt group.
• hunt-tag—Unique sequence number that identifies this
Example: hunt group during configuration tasks. Range: 1 to 100.
Router(config)# ephone-hunt 23 peer
• longest-idle—Calls go to the ephone-dn that has been
idle the longest for the number of hops specified when
the ephone hunt group was defined. The longest-idle is
determined from the last time that a phone registered,
reregistered, or went on-hook.
• peer—First ephone-dn to ring is the number to the right
of the ephone-dn that was the last to ring when the pilot
number was last called. Ringing proceeds in a circular
manner, left to right, for the number of hops specified
when the ephone hunt group was defined.
• sequential—Ephone-dns ring in the left-to-right order
in which they are listed when the hunt group is defined.
Step 4 pilot number [secondary number] Defines the pilot number, which is the number that callers
dial to reach the hunt group.
Example: • number—E.164 number up to 27 characters. The
Router(config-ephone-hunt)# pilot 5601 dialplan pattern can be applied to the pilot number.
• secondary—(Optional) Defines an additional pilot
number for the ephone hunt group.
Step 5 list number[, number...] Defines the list of numbers (from 2 and 20) to which the
ephone hunt group redirects the incoming calls.
Example: • number—E.164 number up to 27 characters. Primary or
Router(config-ephone-hunt)# list 5001, 5002, secondary number assigned to an ephone-dn.
5017, 5028
Example:
Router(config-ephone-hunt)# exit
Step 20 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# exit
Step 24 ephone-dn dn-tag (Optional) Enters ephone-dn configuration mode.
• dn-tag—Tag number for the ephone-dn to be
Example: authorized to join and leave ephone hunt groups.
Router(config)# ephone-dn 29
Step 25 ephone-hunt login (Optional) Enables this ephone-dn to join and leave ephone
hunt groups (dynamic membership).
Example:
Router(config-ephone-dn)# ephone-hunt login
Step 26 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
ephone-hunt 1 longest-idle
pilot 500
list 502, 503, *
max-timeout 30
timeout 10, 10, 10
hops 2
from-ring
fwd-final orig-phone
!
!
ephone-hunt 2 sequential
pilot 600
list 621, *, 623
final 5255348
max-timeout 10
timeout 20, 20, 20
fwd-final orig-phone
!
!
ephone-hunt 77 longest-idle
from-ring
pilot 100
list 101, *, 102
!
Step 2 To verify the configuration of ephone hunt group dynamic membership, use the show running-config
command. Look at the ephone-hunt portion of the output to ensure at least one wildcard slot is
configured. Look at the ephone-dn section to see whether particular ephone-dns are authorized to join
ephone hunt groups. Look at the telephony-service section to see whether FACs are enabled.
Router# show running-config
ephone-hunt 1 longest-idle
pilot 500
list 502, 503, *
max-timeout 30
timeout 10, 10, 10
hops 2
from-ring
fwd-final orig-phone
!
!
ephone-dn 2 dual-line
number 126
preference 1
call-forward busy 500
ephone-hunt login
!
telephony-service
fac custom alias 5 *5 to *35000
fac custom ephone-hunt cancel #5
Step 3 Use the show ephone-hunt command for detailed information about hunt groups, including dial-peer
tag numbers, hunt-group agent status, and on-hook time stamps. This command also displays the
dial-peer tag numbers of all ephone-dns that have joined dynamically and are members of the group at
the time that the command is run.
Router# show ephone-hunt
Group 1
type: peer
pilot number: 450, peer-tag 20123
list of numbers:
451, aux-number A450A0900, # peers 5, logout 0, down 1
peer-tag dn-tag rna login/logout up/down
[20122 42 0 login up ]
[20121 41 0 login up ]
[20120 40 0 login up ]
[20119 30 0 login up ]
[20118 29 0 login down]
452, aux-number A450A0901, # peers 4, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20127 45 0 login up ]
[20126 44 0 login up ]
[20125 43 0 login up ]
[20124 31 0 login up ]
453, aux-number A450A0902, # peers 4, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20131 48 0 login up ]
[20130 47 0 login up ]
[20129 46 0 login up ]
[20128 32 0 login up ]
477, aux-number A450A0903, # peers 1, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20132 499 0 login up ]
preference: 0
preference (sec): 7
timeout: 3, 3, 3, 3
max timeout : 10
hops: 4
next-to-pick: 1
E.164 register: yes
auto logout: no
stat collect: no
Group 2
type: sequential
pilot number: 601, peer-tag 20098
list of numbers:
123, aux-number A601A0200, # peers 1, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20097 56 0 login up ]
622, aux-number A601A0201, # peers 3, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20101 112 0 login up ]
[20100 111 0 login up ]
[20099 110 0 login up ]
623, aux-number A601A0202, # peers 3, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20104 122 0 login up ]
[20103 121 0 login up ]
[20102 120 0 login up ]
*, aux-number A601A0203, # peers 1, logout 0, down 1
peer-tag dn-tag rna login/logout up/down
[20105 0 0 - down]
*, aux-number A601A0204, # peers 1, logout 0, down 1
Prerequisites
Directory numbers to be added to a hunt group must be configured in Cisco Unified CME. For
configuration information, see “SIP: Creating Directory Numbers” on page 162.
Restrictions
• SIP-to-H.323 calls are not supported.
• If call forward is configured for a hunt group member, call forward is ignored by the hunt group.
• Forwarding or transferring to a voice hunt group is not supported.
• Voice-class with codec list can be configured under voice register pool, and more than one list
member will not be supported for B2BUA call.
• Caller ID update is not supported for supplementary services.
• 100 voice hunt groups is the maximum number of hunt group supported.
• Voice hunt groups are subject to max-redirect restriction.
• A pilot dial peer cannot be used as a voice hunt group and a hunt group at the same time.
• If call-waiting is enabled (the default), parallel hunt groups support multiple calls up to the limit of
call-waiting calls supported by the particular SIP phone model. If call waiting is disabled, parallel
hunt groups support only one call at a time in the ringing state. Phones that fail to connect must
return to the on-hook state before they can receive other calls.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice hunt-group hunt-tag [longest-idle | parallel | peer | sequential]
4. pilot number [secondary number]
5. list dn-number, dn-number[, dn-number...]
6. final final-number
7. preference preference-order [secondary secondary-order]
8. hops number
9. timeout seconds
10. end
DETAILED STEPS
Example:
Router# configure terminal
Restrictions
In Cisco Unified CME 4.0 and later, silent ringing, configured on the phone by using the s keyword with
the button command, is suppressed when used with the night service feature. Silent ringing is
overridden and the phone audibly rings during designated night-service periods.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. night-service day day start-time stop-time
5. night-service date month date start-time stop-time
6. night-service everyday start-time stop-time
7. night-service weekday start-time stop-time
8. night-service weekend start-time stop-time
9. night-service code digit-string
10. exit
11. ephone-dn dn-tag
12. night-service bell
13. exit
14. ephone phone-tag
15. night-service bell
16. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# telephony-service
Step 4 night-service day day start-time stop-time Defines a recurring time period associated with a day of the
week during which night service is active.
Example: • day—Day of the week abbreviation. The following are
Router(config-telephony)# night-service day mon valid day abbreviations: sun, mon, tue, wed, thu, fri,
19:00 07:00 sat.
• start-time stop-time—Beginning and ending times for
night service, in an HH:MM format using a 24-hour
clock. If the stop time is a smaller value than the start
time, the stop time occurs the day following the start
time. For example, “mon 19:00 07:00” means “from
Monday at 7 p.m. until Tuesday at 7 a.m.”
Step 5 night-service date month date start-time Defines a recurring time period associated with a month and
stop-time date during which night service is active.
• month—Month abbreviation. The following are valid
Example: month abbreviations: jan, feb, mar, apr, may, jun, jul,
Router(config-telephony)# night-service date aug, sep, oct, nov, dec.
jan 1 00:00 00:00
• date—Date of the month. Range is 1 to 31.
• start-time stop-time—Beginning and ending times for
night service, in an HH:MM format using a 24-hour
clock. The stop time must be greater than the start time.
The value 24:00 is not valid. If 00:00 is entered as a
stop time, it is changed to 23:59. If 00:00 is entered for
both start time and stop time, calls are blocked for the
entire 24-hour period on the specified date.
Step 6 night-service everyday start-time stop-time Defines a recurring night-service time period to be effective
everyday.
Example: • start-time stop-time—Beginning and ending times for
Router(config-telephony)# night-service night service, in an HH:MM format using a 24-hour
everyday 1200 1300 clock. If the stop time is a smaller value than the start
time, the stop time occurs the day following the start
time. For example, “19:00 07:00” means “from 7 p.m.
to 7 a.m. the next morning.” The value 24:00 is not
valid. If 00:00 is entered as a stop time, it is changed to
23:59. If 00:00 is entered for both start time and stop
time, the night service feature will be activated for the
entire 24-hour period.
Example:
Router(config)# telephony-service
Step 4 night-service day day start-time stop-time Defines a recurring time period associated with a day of the
week during which night service is active.
Example: • day—Day of the week abbreviation. The following are
Router(config-telephony)# night-service day mon valid day abbreviations: sun, mon, tue, wed, thu, fri,
19:00 07:00 sat.
• start-time stop-time—Beginning and ending times for
night service, in an HH:MM format using a 24-hour
clock. If the stop time is a smaller value than the start
time, the stop time occurs the day following the start
time. For example, “mon 19:00 07:00” means “from
Monday at 7 p.m. until Tuesday at 7 a.m.”
Step 5 night-service date month date start-time Defines a recurring time period associated with a month and
stop-time date during which night service is active.
• month—Month abbreviation. The following are valid
Example: month abbreviations: jan, feb, mar, apr, may, jun, jul,
Router(config-telephony)# night-service date aug, sep, oct, nov, dec.
jan 1 00:00 00:00
• date—Date of the month. Range is 1 to 31.
• start-time stop-time—Beginning and ending times for
night service, in an HH:MM format using a 24-hour
clock. The stop time must be greater than the start time.
The value 24:00 is not valid. If 00:00 is entered as a
stop time, it is changed to 23:59. If 00:00 is entered for
both start time and stop time, calls are blocked for the
entire 24-hour period on the specified date.
Step 6 night-service everyday start-time stop-time Defines a recurring night-service time period to be effective
everyday.
Example: • start-time stop-time—Beginning and ending times for
Router(config-telephony)# night-service night service, in an HH:MM format using a 24-hour
everyday 1200 1300 clock. If the stop time is a smaller value than the start
time, the stop time occurs the day following the start
time. For example, “19:00 07:00” means “from 7 p.m.
to 7 a.m. the next morning.” The value 24:00 is not
valid. If 00:00 is entered as a stop time, it is changed to
23:59. If 00:00 is entered for both start time and stop
time, the night service feature will be activated for the
entire 24-hour period.
Example:
Router(config-telephony)# exit
Step 11 ephone-dn dn-tag Enters ephone-dn configuration mode to define an
ephone-dn to receive night-service treatment.
Example: • dn-tag—Unique sequence number that identifies the
Router(config)# ephone-dn 55 ephone-dn to receive night-service treatment.
Step 12 night-service bell Marks this ephone-dn for night-service treatment. Incoming
calls to this ephone-dn during the night-service time period
send an alert notification to all IP phones that are marked to
Example:
Router(config-ephone-dn)# night-service bell
receive night-service bell notification.
Example:
Router(config-ephone-dn)# exit
Step 14 ephone phone-tag Enters ephone configuration mode. This is a phone that will
be notified when an incoming call is received by a
night-service ephone-dn during a night-service period.
Example:
Router(config)# ephone 12 • phone-tag—The unique sequence number of the phone
that you are designating as a night-service phone.
Step 15 night-service bell Marks this phone to receive night-service bell notification
when incoming calls are received on ephone-dns marked for
night service during the night-service time period. The alert
Example:
Router(config-ephone)# night-service bell
notification is a splash ring that is not associated with any
of the individual lines on the IP phone and a visual display
of the ephone-dn line number. The phone user can pick up
the call by executing a PickUp or GPickUp.
Step 16 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
telephony-service
fxo hook-flash
load 7910 P00403020214
load 7960-7940 P00303020214
max-ephones 48
max-dn 288
ip source-address 10.50.50.1 port 2000
application segway0
caller-id block code *321
create cnf-files version-stamp 7960 Mar 07 2003 11:19:18
voicemail 79000
max-conferences 8
call-forward pattern .....
moh minuet.wav
date-format yy-mm-dd
transfer-system full-consult
transfer-pattern .....
secondary-dialtone 9
night-service code *1234
night-service day Tue 00:00 23:00
night-service day Wed 01:00 23:59
!
!
CONFIG (Version=4.0(0))
=====================
Version 4.0(0)
Cisco Unified CallManager Express
For on-line documentation please see:
www.cisco.com/en/US/products/sw/voicesw/tsd_products_support_category_home.html
Step 2 Use the show running-config command to verify that the correct ephone-dns and ephones are
configured with the night-service bell command. You can also use the show telephony-service
ephone-dn and show telephony-service ephone commands to display these parameters.
Router# show running-config
ephone-dn 24 dual-line
number 2548
description FrontDesk
night-service bell
ephone 1
mac-address 110F.80C0.FE0B
type 7960 addon 1 7914
no dnd feature-ring
keep-conference
button 1f40 2f41 3f42 4:30
button 7m20 8m21 9m22 10m23
button 11m24 12m25 13m26
night-service bell
Restrictions
• Call waiting is disabled when you configure ephone-dn overlays using the o keyword with the
button command. To enable call waiting, you must configure ephone-dn overlays using the
c keyword with the button command.
• Rollover of overlay calls to another phone button by using the x keyword with the button command
only works to expand coverage if the overlay button is configured with the o keyword in the button
command. Overlay buttons with call waiting that use the c keyword in the button command are not
eligible for overlay rollover.
• In Cisco Unified CME 4.0(3), the Cisco Unified IP Phone 7931G cannot support overlays that
contain ephone-dn configured for dual-line mode.
• The primary ephone-dn on each phone in a shared-line overlay set should be an ephone-dn that is
unique to the phone to guarantee that the phone will have a line available for outgoing calls, and to
ensure that the phone user can obtain dial-tone even when there are no idle lines available in the rest
of the shared-line overlay set. Use a unique ephone-dn in this manner to provide for a unique calling
party identity on outbound calls made by the phone so that the called user can see which specific
phone is calling.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number
5. preference preference-value
6. huntstop
or
no huntstop
7. call-forward noan
8. call-forward busy
9. huntstop channel
10. exit
11. ephone phone-tag
12. mac-address mac-address
13. button button-number{o | c}dn-tag,dn-tag[,dn-tag...] button-number{x}overlay-button-number
14. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn phone-tag [dual-line] Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco Unified IP phone line.
Example: • For shared-line overlay set: Primary ephone-dn on a
Router(config)# ephone-dn 10 dual-line phone should be an ephone-dn that is unique to the
phone.
Step 4 number number Associates a telephone or extension number with the
ephone-dn.
Example:
Router(config-ephone-dn)# number 1001
Step 5 preference preference-order Sets dial-peer preference order for an ephone-dn.
• preference-order—Preference order for the primary
Example: number associated with an extension (ephone-dn). Type
Router(config-ephone-dn)# preference 1 ? for a range of numeric options, where 0 is the highest
preference. Default: 0.
Example:
Router(config-ephone-dn)# exit
Step 11 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone to which you are adding an overlay set.
Router(config)# ephone 4
Step 12 mac-address mac-address Specifies the MAC address of the registering phone.
Example:
Router(config-ephone)# mac-address
1234.5678.abcd
Example:
Router(config-ephone)# end
ephone 5
description Cashier1
mac-address 0117.FBC6.1985
type 7960
button 1o4,5,6,200,201,202,203,204,205,206 2x1 3x1
Step 2 Use the show ephone overlay command to display the configuration and current status of registered
overlay ephone-dns.
Router# show ephone overlay
Step 3 Use the show dialplan number command to display all the number resolutions of a particular phone
number, which allows you to detect whether calls are going to unexpected destinations. This command
is useful for troubleshooting cases in which you dial a number but the expected phone does not ring.
ephone-dn 2
number 5001
preference 2
call-forward busy 6000
call-forward noan 6000
ephone 4
button 1:1 2:2
mac-address 0030.94c3.8724
ephone-dn 11 dual-line
number 1001
no huntstop
huntstop channel
preference 1
ephone-dn 12 dual-line
number 1001
no huntstop
huntstop channel
preference 2
The following example globally disables directed call pickup and changes the action of the PickUp soft
key to perform local group call pickup rather than directed call pickup.
telephony-service
no service directed-pickup
ephone-dn 11
no call-waiting beep accept
number 4411
ephone-dn 12
no call-waiting beep generate
number 4412
telephony-service
max-redirect 8
ephone-dn 24
number 4568
ephone-hunt login
ephone-dn 25
number 4569
ephone-hunt login
ephone-dn 26
number 4570
ephone-hunt login
ephone-hunt 1 peer
list 4566,*,*
timeout 10
final 7777
telephony-service
fac standard
telephony-service
hunt-group logout HLog
fac standard
ephone-template 7
softkeys connected Endcall Hold Transfer HLog
softkeys idle Newcall Redial Pickup Cfwdall HLog
softkeys seized Endcall Redial Pickup Cfwdall HLog
telephony-service
hunt-group logout HLog
The following example enables automatic status change to not-ready after two unanswered hunt group
calls for any ephone-dn that dynamically logs in to the hunt group using the wildcard slot in the hunt
group list. Phones that are automatically placed in the not-ready status when they do not answer two
hunt-group calls are also placed into DND status (they will also not accept directly dialed calls).
ephone-hunt 3 peer
pilot 4200
list 1001, 1002, *
timeout 10
auto logout 2 dynamic
final 4500
telephony-service
hunt-group logout DND
telephony-service
night-service day mon 17:00 08:00
night-service day tue 17:00 08:00
night-service day wed 17:00 08:00
night-service day thu 17:00 08:00
night-service day fri 17:00 08:00
night-service day sat 13:00 12:00
night-service day sun 12:00 08:00
night-service code *6483
!
ephone-dn 1
number 1000
night-service bell
!
ephone-dn 2
number 1001
night-service bell
!
ephone-dn 10
number 2222
!
ephone-dn 11
number 3333
!
ephone 5
mac-address 1111.2222.0001
button 1:1 2:2
!
ephone 14
mac-address 1111.2222.0002
button 1:10
night-service bell
!
ephone 15
mac-address 1111.2222.0003
button 1:11
night-service bell
ephone-dn 2
number 1001
no huntstop
preference 1
ephone-dn 3
number 1001
huntstop
preference 2
ephone 10
button 1o1,2,3
ephone 11
button 1o1,2,3
ephone 12
button 1o1,2,3
ephone-dn 11 dual-line
number 1001
no huntstop
huntstop channel
preference 1
ephone-dn 12 dual-line
number 1001
no huntstop
huntstop channel
preference 2
ephone-dn 13 dual-line
number 1001
preference 3
no huntstop
huntstop channel
ephone-dn 14 dual-line
number 1001
preference 4
huntstop
huntstop channel
ephone 33
mac 00e4.5377.2a33
button 1o10,11,12,13,14
ephone 34
mac 9c33.0033.4d34
button 1o10,11,12,13,14
ephone 35
mac 1100.8c11.3865
button 1o10,11,12,13,14
ephone 36
mac 0111.9c87.3586
button 1o10,11,12,13,14
ephone 37
mac 01a4.8222.3911
button 1o10,11,12,13,14
huntstop-channel
!
!The following ephone configuration includes (unique) ephone-dn 1 as the primary line in a
shared-line overlay
ephone 1
mac-address 1111.1111.1111
button 1o1,10,11,12
!
!The next ephone configuration includes (unique) ephone-dn 2 as the primary line in
another shared-line overlay
!
ephone 2
mac-address 2222.2222.2222
button 1o2,10,11,12
ephone-dn 2 dual-line
number 1001
ephone-dn 3 dual-line
number 1001
ephone-dn 10 dual-line
number 1111
no huntstop
huntstop channel
call-forward noan 7000 timeout 30
ephone-dn 11 dual-line
number 1111
preference 1
no huntstop
huntstop channel
call-forward noan 7000 timeout 30
ephone-dn 12 dual-line
number 1111
preference 2
huntstop channel
call-forward noan 7000 timeout 30
call-forward busy 7000
ephone 1
button 1c1,10,11,12
ephone 2
button 1c2,10,11,12
ephone 3
button 1c3,10,11,12
ephone-dn 12
number 2012
ephone-dn 13
number 2013
ephone-dn 14
number 2014
.
.
.
ephone-dn 28
number 2028
ephone 1
button 1o11,12,13,20,21,22,23,24,25,26,27,28 2x1 3x1
ephone 2
button 1o14,15,16,20,21,22,23,24,25,26,27,28 2x1 3x1
ephone 3
button 1o17,18,19,20,21,22,23,24,25,26,27,28 2x1 3x1
telephony-service
service dnis dir-lookup
ephone-dn 1
number 5500 secondary 555000.
ephone-dn 2
number 5501 secondary 555001.
ephone-dn 3
number 5502 secondary 555002.
ephone 1
button 1o1,2,3
mac-address 1111.1111.1111
ephone 2
button 1o1,2,3
mac-address 2222.2222.2222
ephone 3
button 1o1,2,3
mac-address 3333.3333.3333
The following example shows a hunt-group configuration for a medical answering service with two
phones and four doctors. Each phone has two buttons, and each button is assigned two doctors’ numbers.
When a patient calls 5550341, Cisco Unified CME matches the hunt-group pilot secondary number
(555....), rings button 1 on one of the two phones, and displays “doctor1.” For more information about
hunt-group behavior, see the “Hunt Groups” section on page 569. Note that wildcards are used only in
secondary numbers and cannot be used with primary numbers.
telephony-service
service dnis dir-lookup
max-redirect 20
directory entry 1 5550341 name doctor1
directory entry 2 5550772 name doctor1
directory entry 3 5550263 name doctor3
directory entry 4 5550150 name doctor4
ephone-dn 1
number 1001
ephone-dn 2
number 1002
ephone-dn 3
number 1003
ephone-dn 4
number 104
ephone 1
button 1o1,2
button 2o3,4
mac-address 1111.1111.1111
ephone 2
button 1o1,2
button 2o3,4
mac-address 2222.2222.2222
ephone-hunt 1 peer
pilot number 5100 secondary 555....
list 1001, 1002, 1003, 1004
final number 5556000
hops 5
preference 1
timeout 20
no-reg
ephone-dn 1
number 18005550100
ephone-dn 2
name catalog1
number 18005550101
ephone-dn 3
name catalog2
number 18005550102
ephone-dn 4
name catalog3
number 18005550103
ephone 1
button 1o1,2,3,4
ephone 2
button 1o1,2,3,4
ephone 3
button 1o1,2,3,4
Where to Go Next
Dial-Peer Call Hunt and Hunt Groups
Dial peers other than ephone-dn dial peers can be directly configured as hunt groups or rotary groups,
in which multiple dial peers can match incoming calls. (These are not the same as Cisco Unified CME
ephone hunt groups.) For more information, see the “Hunt Groups” section of the “Dial Peers Features
and Configuration” chapter of Dial Peer Configuration on Voice Gateway Routers.
Called-Name Display
This feature allows you to specify that the name of the called party, rather than the number, should be
displayed for incoming calls. This feature is very helpful for agents answering calls for multiple
ephone-dns that appear on a single line button in an ephone-dn overlay set. For more information, see
“Configuring Directory Services” on page 689.
Ephone-dn Templates
The ephone-hunt login command authorizes an ephone-dn to dynamically join and leave an ephone
hunt group. It can be included in an ephone-dn template that is applied to one or more individual
ephone-dns. For more information, see “Creating Templates” on page 881.
Do Not Disturb
The Do Not Disturb (DND) feature can be used as an alternative to the HLog function for preventing
incoming calls from ringing on a phone. The difference is that HLog prevents only hunt group calls from
ringing, while DND prevents all calls from ringing. For more information, see “Configuring Do Not
Disturb” on page 709.
Ephone Templates
The night-service bell command specifies that a phone will receive night-service notification when calls
are received at ephone-dns configured as night-service ephone-dns. This command can be included in
an ephone template that is applied to one or more individual ephones.
For more information, see “Creating Templates” on page 881.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 31 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the caller-ID (CLID) blocking feature in Cisco Unified Communications
Manager Express (Cisco Unified CME).
Contents
• Restrictions for Caller ID Blocking, page 639
• Information about Caller ID Blocking, page 639
• How to Configure Caller ID Blocking, page 640
• Configuration Examples for Caller ID Blocking, page 644
• Additional References, page 644
• Feature Information for Caller ID Blocking, page 646
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. caller-id block code code-string
5. exit
6. ephone-dn dn-tag
7. caller-id block
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 caller-id block code code-string (Optional) Defines a code that users can enter before making
calls on which the caller ID should not be displayed.
Example: • code-string—Digit string of up to 16 characters. The first
Router(config-telephony)# caller-id block character must be an asterisk (*).
code *1234
Step 5 exit Exits telephony-service configuration mode.
Example:
Router(config-telephony)# exit
Step 6 ephone-dn dn-tag Enters ephone-dn configuration mode.
Example:
Router(config)# ephone-dn 3
Step 7 caller-id block (Optional) Blocks display of all caller-ID information for
outbound calls that originate from this ephone-dn.
Example: • By default, caller ID is not blocked on calls that originate
Router(config-ephone-dn)# caller-id block from a Cisco Unified IP phone.
• This command tells the far-end gateway device to block
display of calling-party information for the calls received
from this ephone-dn.
Step 8 end Returns to privileged EXEC mode.
Example:
Router(config-dial-peer)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag [pots | voip]
4. clid strip
5. clid strip name
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dial-peer voice tag [pots | voip] Enters dial-peer configuration mode.
Note You can configure caller-ID blocking on POTS dial
Example: peers if the POTS interface is ISDN. This feature is not
Router(config)# dial-peer voice 3 voip available on FXO/CAS lines.
Step 4 clid strip (Optional) Removes the calling-party number from the CLID
information being sent with VoIP calls.
Example:
Router(config-dial-peer)# clid strip
Step 5 clid strip name (Optional) Removes the calling-party name from the CLID
information being sent with VoIP calls.
Example:
Router(config-dial-peer)# clid strip name
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-dial-peer)# end
The following example blocks the display of CLID name and number on VoIP calls but allows CLID
display for local calls:
ephone-dn 3
number 2345
dial-peer voice 2 voip
clid strip
clid strip name
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 32 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the conferencing support in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Restrictions for Conferencing, page 647
• Information About Conferencing, page 648
• How to Configure Conferencing, page 651
• Configuration Examples for Conferencing, page 671
• Where to Go Next, page 686
• Additional References, page 686
• Feature Information for Conferencing, page 687
Conferencing Overview
Conferencing allows you to join three or more parties in a telephone conversation. Two types of
conferencing are available in Cisco Unified CME: ad hoc and meet-me.
Ad hoc conferences are created when one party calls another party, then either party adds one or more
parties to the conference call. Ad hoc conferences can be hardware-based or software-based.
Hardware-based ad hoc conferencing uses digital signal processors (DSPs) to enhance ad hoc
conferencing by allowing more parties than software-based ad hoc conferencing, which allows three
parties only.
Meet-me conferences are created by parties calling a designated conference number. Meet-me
conferencing is hardware-based only. If you configure software-based conferencing, you cannot have
meet-me conferences.
Ad Hoc Conferencing
Before Cisco Unified CME 4.1, support for conferencing is limited to three-party ad hoc conference
calls using a G.711 codec. To have an ad hoc conference with a party that is not using a G.711 codec,
transcoding is necessary. For more information, see the “Transcoding When a Remote Phone Uses
G.729r8” section on page 305.
The maximum number of simultaneous conferences is platform-specific to the type of
Cisco Unified CME router, and each individual Cisco Unified IP phone can host a maximum of one
conference at a time. You cannot create a second conference on a phone if you already have an existing
conference on hold.
End-of-Conference Options
For Cisco CME 3.2 and later versions, a person who initiates a conference call and hangs up can either
keep the remaining parties connected or disconnect them.
Cisco Unified IP phones can be configured to keep the remaining conference parties connected when the
conference initiator hangs up (places the handset back in the on-hook position). Conference originators
can disconnect from their conference calls by pressing the Confrn (conference) soft key. When an
initiator uses the Confrn key to disconnect from the conference call, the oldest call leg will be put on
hold, leaving the initiator connected to the most recent call leg. The conference initiator can then
navigate between the two parties by pressing either the Hold soft key or the line buttons to select the
desired call.
In Cisco Unified CME 4.0 and later versions, behavior for the end of three-way conferences can be
configured at a phone level. The options specify whether the last party that joined a conference can be
dropped from the conference and whether the remaining two parties should be allowed to continue their
connection after the conference initiator has left the conference.
x1215 x1225
1
IP IP
x1235
170458
IP
You can configure ad hoc conferencing so that only the creator can add parties to the conference. The
default is that any party can add other parties to the conference.
You can configure conferencing so that the conference drops when the creator hangs up, and you can
configure it so that the conference drops when the last local party hangs up. The default is that the
conference is not dropped, regardless of whether the creator hangs up, provided three parties remain in
the conference.
For configuration information, see the “SCCP: Configuring Conferencing Options for a Phone” section
on page 667 for more information.
Meet-Me Conferencing
In Cisco Unified CME 4.1 and later versions, meet-me conferences consist of at least three parties
dialing a meet-me conference number predetermined by a system administrator. For example, the
conference shown in Figure 42 is created when the conference creator at extension 1215 presses the
MeetMe soft key and hears a confirmation tone, then dials the meet-me conference number 1500.
Extension 1225 and extension 1235 join the meet-me conference by dialing 1500. Extensions 1215,
1225, and 1235 are now parties in a meet-me conference on extension 1500.
x1500
1 3 2
IP IP
x1215 x1225
170459
IP
x1235
• ConfList—Conference list. Lists all parties in a conference. For ad hoc conferences, this soft key is
available for all parties in a conference. For meet-me conferences, this soft key is available for the
creator only. Press Update to update the list of parties in the conference, for instance, to verify that
a party has been removed from the conference.
• Join—Joins an established call to conference. After you press Select to choose an established call
or conference, press Join to join that call or conference to the established call or conference.
• RmLstC—Remove last caller. Removes the last party added to the conference. This soft key works
for the creator only.
• Select—Selects a call or conference to join to conference and selects a call to remove from a
conference. The creator can remove other parties by pressing the ConfList soft key, then use the
Select and Remove soft keys to remove the appropriate parties.
• MeetMe—Initiates a meet-me conference. This soft key is pressed by the creator before dialing the
conference number. Other meet-me conference parties dial the conference number only to join the
conference. This soft key must be configured before you can initiate meet-me conferences.
Three-Party Ad Conferencing
• Modifying the Default Configuration for Three-Party Ad Hoc Conferencing, page 651 (optional)
• SCCP: Configuring Conferencing Options on a Phone, page 653 (optional)
• SIP: Configuring Conferencing Options on a Phone, page 655 (optional)
Restrictions
• When a three-way conference is established, a participant cannot use call transfer to join the
remaining conference participants to a different number.
• Three-party ad hoc conferencing does not support meet-me conferences.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. max-conferences max-conference-number [gain -6 | 0 | 3 | 6]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)#
Step 4 max-conferences max-conference-number Sets the maximum number of simultaneous three-party
[gain -6 | 0 | 3 | 6] conferences supported by the router.
• max-conference-number—Maximum number of
Example: simultaneous three-party conferences supported by a router,
Router(config-telephony)# max-conferences which is platform-dependent. The default is half of the
6
maximum number. The maximum number of conferences per
platform is as follows:
– Cisco 2600 series, Cisco 2801—8
– Cisco 2811, Cisco 2821, Cisco 2851, Cisco 3600 series,
Cisco 3700 series—16
– Cisco 3800 series—24 (requires Cisco IOS
Release 12.3(11)XL or a higher release)
• gain—(Optional) Amount to increase the sound volume of
VoIP and PSTN calls joining a conference call, in decibels.
Valid values are -6, 0, 3, and 6. The default is -6.
Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-telephony)# end
What to Do Next
• To configure optional end-of-conference options for three-party ad hoc conferencing on SCCP
phones, see “SCCP: Configuring Conferencing Options on a Phone” section on page 653
• To configure optional end-of-conference options for three-party ad hoc conferencing on SCCP
phones, see “SIP: Configuring Conferencing Options on a Phone” section on page 655
Prerequisites
• Conferencing uses call transfer to connect the two remaining parties of a conference when a
conference initiator leaves the conference. To use this feature, you must configure the
transfer-system command. For configuration information, see “Configuring Call Transfer and
Forwarding” on page 499.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. keep-conference [drop-last] [endcall] [local-only]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 1
Example:
Router(config)# end
What to Do Next
If you are finished modifying the configuration, you are ready to generate configuration files for the
phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones” on page 247.
Prerequisites
• To facilitate call transfer by using the Confrn softkey, conference and transfer attended or transfer
blind must be enabled. For configuration information, see “Configuring Call Transfer and
Forwarding” on page 499.
Restrictions
Music on hold (MOH) is not supported for call hold invoked from a SIP phone. A caller hears only
silence when placed on hold by a SIP phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. keep-conference
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example: • pool-tag unique sequence number of the SIP phone to
Router(config)# voice register pool 3 be configured. Range is 1 to 100 or the upper limit as
defined by max-pool command.
• Default is enabled.
• Remaining calls are transferred without consultation as
enabled by the transfer-attended (voice register
template) or transfer-blind (voice register template)
commands.
Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-register-pool)# end
What to Do Next
• If you are finished modifying the configuration, you are ready to generate configuration files for the
phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones” on page 250.
Step 1 Use the debug ephone commands to observe messages and states associated with an ephone. For more
information, see the Cisco Unified CME Command Reference.
Prerequisites
• Cisco Unified CME 4.1 or a later version
• You must have a PVDM2-8, PVDM2-16, PVDM2-32, or PVDM2-64 high-density packet voice
digital signal processor module hosted on the motherboard or on a module such as the NM-HDV2
or NM-HD-2VE.
• For Cisco Unified IP Phone 7985, firmware version 4-1-2-0 or a later version
Restrictions
• The maximum number of meet-me conference parties is 32 for one DSP using the G.711 codec and
16 for the G.729 codec.
• A participant cannot join more than one conference at the same time.
• Ad hoc conferencing for more than three parties is not supported on the Cisco Unified IP Phone
7906 and 7910 and Cisco Unified IP Phone 7914 Expansion Module.
• Ad hoc conferencing for more than three parties is not supported on Cisco Unified IP phones
running SIP.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice-card slot
4. dsp services dspfarm
5. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice-card slot Enters voice-card configuration mode and configure a voice
card.
Example:
Router(config)# voice-card 2
Step 4 dsp services dspfarm Enables digital-signal-processor (DSP) farm services for a
particular voice network module.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 5 exit Exits voice-card configuration mode.
Example:
Router(config-voicecard)# exit
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice class custom-cptone cptone-name Creates a voice class for defining custom call-progress
tones to be detected.
Example:
Router(config)# voice class custom-cptone
jointone
Step 4 dualtone conference Configures conference join and leave tones.
Example:
Router(cfg-cptone)# dualtone conference
Step 5 frequency frequency-1 [frequency-2] Defines the frequency components for a call-progress tone.
Example:
Router(cfg-cp-dualtone)# frequency 600 900
Step 6 cadence {cycle-1-on-time cycle-1-off-time Defines the tone-on and tone-off durations for a
[cycle-2-on-time cycle-2-off-time] call-progress tone.
[cycle-3-on-time cycle-3-off-time]
[cycle-4-on-time cycle-4-off-time]} |
continuous
Example:
Router(cfg-cp-dualtone)# cadence 300 150 300
100 300 50
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(cfg-cp-dualtone)# exit
SUMMARY STEPS
1. enable
2. configure terminal
3. sccp local interface-type interface-number [port port-number]
4. sccp ccm {ip-address | dns} identifier identifier-number [priority priority] [port port-number]
[version version-number]
5. sccp ccm group group-number
6. bind interface interface-type interface-number
7. exit
8. sccp
9. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 sccp local interface-type interface-number Selects the local interface that SCCP applications
[port port-number] (transcoding and conferencing) use to register with
Cisco Unified CME.
Example:
Router(config)# sccp local FastEthernet0/0
Step 4 sccp ccm {ip-address | dns} identifier Adds a Cisco Unified CME router to the list of available
identifier-number [priority priority] [port servers and set various parameters—including IP address or
port-number] [version version-number]
Domain Name System (DNS) name, port number, and
version number.
Example: • version-number—Must be 4.0 or later.
Router(config)# sccp ccm 1.4.158.3 identifier
100 version 4.0
Step 5 sccp ccm group group-number Creates a Cisco Unified CME group.
Example:
Router(config)# sccp ccm group 123
Example:
Router(config-sccp-cm)# bind interface
fastethernet 0/0
Step 7 exit Exits SCCP Cisco Unified CME configuration mode.
Example:
Router(config-sccp-cm)# exit
Step 8 sccp Enables SCCP and its related applications (transcoding and
conferencing).
Example:
Router(config)# sccp
Step 9 exit Exits global configuration mode.
Example:
Router(config)# exit
Note The DSP farm can be on the same router as the Cisco Unified CME or on a different router.
SUMMARY STEPS
1. enable
2. configure terminal
3. dspfarm profile profile-identifier conference
4. codec {codec-type | pass-through}
5. conference-join custom-cptone cptone-name
6. conference-leave custom-cptone cptone-name
7. maximum conference-party max-parties
8. maximum sessions number
9. associate application sccp
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 dspfadrm profile profile-identifier conference Enters DSP farm profile configuration mode and defines a
profile for DSP farm services.
Example:
Router(config)# dspfarm profile 1 conference
Step 4 codec {codec-type | pass-through} Specifies the codecs supported by a DSP farm profile.
Note Repeat this step as necessary to specify all the
Example: supported codecs.
Router(config-dspfarm-profile)# codec g711ulaw
Step 5 conference-join custom-cptone cptone-name Associates a custom call-progress tone to indicate joining a
conference with a DSP farm profile.
Example: Note The cptone-name argument in this step must be the
Router(config-dspfarm-profile)# conference-join same as the cptone-argument in the voice class
custom-cptone jointone custom-cptone command configured in the “SCCP:
Enabling DSP Farm Services for a Voice Card”
section on page 658.
Step 6 conference-leave custom-cptone cptone-name Associates a custom call-progress tone to indicate leaving a
conference with a DSP farm profile.
Example: Note The cptone-name argument in this step must be the
Router(config-dspfarm-profile)# same as the cptone-argument in the voice class
conference-leave custom-cptone leavetone custom-cptone command configured in the “SCCP:
Enabling DSP Farm Services for a Voice Card”
section on page 658.
Step 7 maximum conference-party max-parties (Optional) Configures the maximum number of conference
parties allowed in each meet-me conference. The maximum
is codec-dependent.
Example:
Router(config-dspfarm-profile)# maximum
conference-party 32
Step 8 maximum sessions number Specifies the maximum number of sessions that are
supported by the profile.
Example:
Router(config-dspfarm-profile)# maximum
sessions 8
Example:
Router(config-dspfarm-profile)# associate
application sccp
Step 10 end Exits to privileged EXEC mode.
Example:
Router(config-dspfarm-profile)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. sccp ccm group group-number
4. associate ccm identifier-number priority priority-number
5. associate profile profile-identifier register device-name
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 sccp ccm group group-number Creates a Cisco Unified CME group.
Example:
Router(config)# sccp ccm group 1
Step 4 associate ccm identifier-number priority Associates a Cisco Unified CME router with the group and
priority-number establishes its priority within the group.
Example:
Router(config-sccp-ccm)# associate ccm 100
priority 1
Example:
Router(config-sccp-ccm)# end
Note Configuring multi-party ad hoc conferencing in Cisco Unified CME disables three-party ad hoc
conferencing.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. conference hardware
5. sdspfarm units number
6. sdspfarm tag number device-name
7. sdspfarm conference mute-on mute-on-digits mute-off mute-off-digits
8. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# telephony-service
Step 4 conference hardware Configures a Cisco Unified CME system for multi-party
conferencing only.
Example:
Router(config-telephony)# conference hardware
Step 5 sdspfarm units number Specifies the maximum number of DSP farms that are
allowed to be registered to the SCCP server.
Example:
Router(config-telephony)# sdspfarm units 3
Step 6 sdspfarm tag number device-name Permits a DSP farm to register to Cisco Unified CME and
associates it with a SCCP client interface's MAC address.
Example: Note The device-name in this step must be the same as the
Router(config-telephony)# sdspfarm tag 2 device-name in the associate profile command in
confdsp1 Step 5 of the “SCCP: Associating Cisco Unified
CME with a DSP Farm Profile” section on
page 663.
Step 7 sdspfarm conference mute-on mute-on-digits Defines mute-on and mute-off digits for conferencing.
mute-off mute-off-digits
• Maximum: 3 digits. Valid values are the numbers and
symbols that appear on your telephone keypad: 1, 2, 3,
Example: 4, 5, 6, 7, 8, 9, 0, *, and #.
Router(config-telephony)# sdspfarm conference
mute-on 111 mute-off 222 • Mute-on and mute-off digits can be the same.
Step 8 end Exits to privileged EXEC mode.
Example:
Router(config-telephony)# end
Note Ensure that you configure enough directory numbers to accommodate the anticipated number of
conferences. The maximum number of parties in a multi-party ad hoc conference on an IP phone is eight;
the maximum on an analog phone is three.
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag dual-line Enters ephone-dn configuration mode for the purposes of
creating and configuring an extension (ephone-dn) for a
Cisco Unified IP phone line.
Example:
Router(config)# ephone-dn 18 dual-line • Each ephone-dn can carry two parties if it is configured
as a dual line.
• Configure enough ephone-dns to accommodate the
maximum number of conference participants to be
supported.
• For multi-party ad hoc conferencing, maximum number
of directory numbers is 8, but you can configure a lower
maximum.
• For meet-me conferencing, maximum number of
directory numbers is 32, but you can configure a lower
maximum.
• Minimum number of directory numbers required: 2.
Step 4 number number [secondary number] [no-reg [both Associates a telephone or extension number with an
| primary]] ephone-dn in a Cisco Unified CME system.
• Each DN for a conference must have the same primary
Example: and secondary number.
Router(config-ephone-dn)# number 6789
Example:
Router(config-ephone-dn)# end
Note The following commands can also be configured in ephone configuration mode. Commands configured
in ephone configuration mode have priority over commands in ephone-template configuration mode.
Restrictions
• The ConfList (including the Remove, Update, and Exit soft keys within the ConfList function) and
RmLstC soft keys do not work on a Cisco Unified IP Phone 7902, 7935, and 7936.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-template template-tag
4. conference add-mode [creator]
5. conference drop-mode [creator | local]
6. conference admin
7. softkeys connected [Acct] [ConfList] [Confrn] [Endcall] [Flash] [HLog] [Hold] [Join] [Park]
[RmLstC] [Select] [Trnsfer]
8. softkeys hold [Join] [Newcall] [Resume] [Select]
9. softkeys idle [Cfwdall] [ConfList] [Dnd] [Gpickup] [HLog] [Join] [Login] [Newcall] [Pickup]
[Redial] [RmLstC]
10. softkeys seized [CallBack] [Cfwdall] [Endcall] [Gpickup] [HLog] [MeetMe] [Pickup] [Redial]
11. exit
12. ephone phone-tag
13. ephone-template template-tag
14. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enter ephone-template configuration mode to create an
ephone template to configure a set of phone features.
Example:
Router(config)# ephone-template 1
Step 4 conference add-mode [creator] (Optional) Configures the mode for adding parties to
conferences.
Example: • creator—Only the creator can add parties to the
Router(config-ephone-template)# conference conference.
add-mode creator
Step 5 conference drop-mode [creator | local] (Optional) Configures the mode for dropping parties from
multi-party ad hoc and meet-me conferences.
Example: • creator—The active conference terminates when the
Router(config-ephone-template)# conference creator hangs up.
drop-mode creator
• local—The active conference terminates when the last
local party in the conference hangs up or drops out of
the conference.
Example:
Router(config-ephone-template)# exit
Step 12 ephone phone-tag Enters Ethernet phone (ephone) configuration mode for an
IP phone for the purposes of creating and configuring an
ephone.
Example:
Router(config)# ephone 1
Example:
Router(config-ephone)# exit
What to Do Next
If you are finished modifying the configuration, you are ready to generate configuration files for the
phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones” on page 247.
Ad-hoc 0 8 A001
DN tags: 13, 14, 15, 16
Meetme 0 8 1234
DN tags: 20, 21, 22, 23
ephone 24
button 1:35
keep-conference drop-last local-only
In the following example, extension 3666 initiates a three-way conference. After the conference is
established, extension 3666 can press the Confrn soft key to disconnect the last party that was connected
and remain connected to the first party that was connected. Also, extension 3666 can hang up or press
the EndCall soft key to leave the conference and keep the other two parties connected.
ephone-dn 36
number 3666
ephone 25
button 1:36
keep-conference drop-last endcall
In the following example, extension 3777 initiates a three-way conference. After the conference is
established, extension 3777 can press the Confrn soft key to disconnect the last party that was connected
and remain connected to the first party that was connected. Also, extension 3777 can hang up or press
the EndCall soft key to leave the conference and keep the other two parties connected only if one of the
two parties is local to the Cisco Unified CME system.
ephone-dn 38
number 3777
ephone 27
button 1:38
keep-conference drop-last endcall local-only
In the following example, extension 3999 initiates a three-way conference. After the conference is
established, extension 3999 can hang up or press the EndCall soft key to leave the conference and keep
the other two parties connected only if one of the two parties is local to the Cisco Unified CME system.
Extension 3999 can also use the Confrn soft key to break up the conference but stay connected to both
parties.
ephone-dn 39
number 3999
ephone 29
button 1:39
keep-conference endcall local-only
DSP Farm and Cisco Unified CME on the Same Router: Example
In this example, the DSP farm and Cisco Unified CME are on the same router as shown in Figure 43.
IP
Cisco Unified CME
DSP farm SIP
WAN
LAN
SCCP FXS VG224 H323 FXS
SCCP
IP
PSTN
IPC
IP link 170540
resource policy
!
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
!
ip dhcp pool phone1
host 10.4.188.66 255.255.0.0
client-identifier 0100.0ab7.b144.4a
default-router 10.4.188.65
option 150 ip 10.4.188.65
!
ip dhcp pool phone2
host 1.4.188.67 255.255.0.0
client-identifier 0100.3094.c269.35
default-router 10.4.188.65
option 150 ip 10.4.188.65
!
!
voice-card 1
dsp services dspfarm
!
!
voice call send-alert
voice call carrier capacity active
!
voice service voip
allow-connections h323 to h323
supplementary-service h450.12
h323
!
!
!
!
controller E1 1/0
framing NO-CRC4
!
controller E1 1/1
!
!
interface FastEthernet0/0
ip address 10.4.188.65 255.255.0.0
duplex auto
speed auto
no keepalive
no cdp enable
no clns route-cache
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
no clns route-cache
!
ip route 10.4.0.0 255.255.0.0 FastEthernet0/0
ip route 192.168.254.254 255.255.255.255 10.4.0.1
!
ip http server
!
!
control-plane
!
!
sccp local FastEthernet0/0
sccp ccm 10.4.188.65 identifier 1 version 4.0
sccp
!
sccp ccm group 123
associate ccm 1 priority 1
associate profile 1 register mtp00097c5e9ce0
keepalive retries 5
!
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 6
associate application SCCP
!
dial-peer cor custom
!
!
!
dial-peer voice 6 voip
destination-pattern 6...
session target ipv4:10.4.188.90
!
telephony-service
conference hardware
load 7960-7940 P00307020400
load 7905 CP7905060100SCCP050309A.sbin
max-ephones 48
max-dn 180
ip source-address 10.4.188.65 port 2000
timeouts ringing 500
system message MY MELODY (2611)
sdspfarm units 4
sdspfarm tag 1 mtp00097c5e9ce0
max-conferences 4 gain -6
call-forward pattern ....
transfer-system full-consult
transfer-pattern 7...
transfer-pattern ....
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1
softkeys hold Newcall Resume Select Join
softkeys idle Cfwdall ConfList Dnd Gpickup HLog Join Login Newcall Pickup Redial RmLstC
softkeys seized Redial Pickup Gpickup HLog Meetme Endcall
softkeys connected Acct ConfList Confrn Endcall Flash HLog Hold Join Park RmLstC Select
Trnsfer
!
!
ephone-dn 1 dual-line
number 8001
name melody-8001
!
!
ephone-dn 2 dual-line
number 8002
!
!
ephone-dn 3 dual-line
number 8003
!
!
ephone-dn 4 dual-line
number 8004
!
!
ephone-dn 5 dual-line
number 8005
!
!
ephone-dn 6 dual-line
number 8006
!
!
ephone-dn 7 dual-line
number 8007
!
!
ephone-dn 8 dual-line
number 8008
!
!
ephone-dn 60 dual-line
number 8887
conference meetme
no huntstop
!
!
ephone-dn 61 dual-line
number 8887
conference meetme
preference 1
no huntstop
!
!
ephone-dn 62 dual-line
number 8887
conference meetme
preference 2
no huntstop
!
!
ephone-dn 63 dual-line
number 8887
conference meetme
preference 3
!
!
ephone-dn 64 dual-line
number 8889
name Conference
conference ad-hoc
no huntstop
!
!
ephone-dn 65 dual-line
number 8889
name Conference
conference ad-hoc
preference 1
no huntstop
!
!
ephone-dn 66 dual-line
number 8889
name Conference
conference ad-hoc
preference 2
no huntstop
!
!
ephone-dn 67 dual-line
number 8889
name Conference
conference ad-hoc
preference 3
!
!
ephone 1
ephone-template 1
mac-address 0030.94C2.6935
type 7960
button 1:1 2:2
!
!
ephone 2
ephone-template 1
mac-address 000A.B7B1.444A
type 7940
button 1:4 2:8
!
line con 0
exec-timeout 0 0
line aux 0
exec-timeout 0 0
line vty 0 4
exec-timeout 0 0
login
line vty 5 15
login
!
!
end
Figure 44 Cisco Unified CME and the DSP Farm on Different Routers
IP
Cisco Unified CME SIP
WAN
LAN
SCCP FXS VG224 H323 FXS
SCCP
IP
PSTN
IPC
PSTN call
IP link
170541
DSP farm
voice-card 1
no dspfarm
!
voice-card 3
dspfarm
!
ip cef
!
!
no ip dhcp use vrf connected
!
ip dhcp pool IPPhones
network 10.15.15.0 255.255.255.0
option 150 ip 10.15.15.1
default-router 10.15.15.1
!
!
interface FastEthernet0/0
ip address 10.3.111.102 255.255.0.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
duplex auto
speed auto
!
interface FastEthernet0/1.1
encapsulation dot1Q 10
ip address 10.15.14.1 255.255.255.0
!
interface FastEthernet0/1.2
encapsulation dot1Q 20
ip address 10.15.15.1 255.255.255.0
!
ip route 0.0.0.0 0.0.0.0 10.5.51.1
ip route 0.0.0.0 0.0.0.0 10.3.0.1
!
ip http server
!
!
!
!
control-plane!
!
!
!
dial-peer voice 1 voip
destination-pattern 3...
session target ipv4:10.3.111.101
!
!
telephony-service
conference hardware
load 7910 P00403020214
load 7960-7940 P003-07-5-00
max-ephones 50
max-dn 200
ip source-address 10.15.15.1 port 2000
sdspfarm units 4
sdspfarm transcode sessions 12
sdspfarm tag 1 confer1
sdspfarm tag 4 xcode1
max-conferences 8 gain -6
moh flash:music-on-hold.au
multicast moh 239.0.0.0 port 2000
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1
softkeys hold Resume Newcall Select Join
softkeys idle Redial Newcall ConfList RmLstC Cfwdall Join Pickup Login HLog Dnd Gpickup
softkeys seized Endcall Redial Cfwdall Meetme Pickup Callback
softkeys alerting Endcall Callback
softkeys connected Hold Endcall Confrn Trnsfer Select Join ConfList RmLstC Park Flash
!
ephone-dn 1 dual-line
number 6000
!
!
ephone-dn 2 dual-line
number 6001
!
!
ephone-dn 3 dual-line
number 6002
!
!
ephone-dn 4 dual-line
number 6003
!
!
ephone-dn 5 dual-line
number 6004
!
!
ephone-dn 6 dual-line
number 6005
!
!
ephone-dn 7 dual-line
number 6006
!
!
ephone-dn 8 dual-line
number 6007
!
!
ephone-dn 9 dual-line
number 6008
!
!
ephone-dn 10 dual-line
number 6009
!
!
ephone-dn 11
number 6011
!
!
ephone-dn 12
number 6012
!
!
ephone-dn 13
number 6013
!
!
ephone-dn 14
number 6014
!
!
ephone-dn 15
number 6015
!
!
ephone-dn 16
number 6016
!
!
ephone-dn 17
number 6017
!
!
ephone-dn 18
number 6018
!
!
ephone-dn 19
number 6019
!
!
ephone-dn 20
number 6020
!
!
ephone-dn 21
number 6021
!
!
ephone-dn 22
number 6022
!
!
ephone-dn 23
number 6023
!
!
ephone-dn 24
number 6024
!
!
ephone-dn 25 dual-line
number 6666
conference meetme
preference 1
no huntstop
!
!
ephone-dn 26 dual-line
number 6666
conference meetme
preference 2
no huntstop
!
!
ephone-dn 27 dual-line
number 6666
conference meetme
preference 3
no huntstop
!
!
ephone-dn 28 dual-line
number 6666
conference meetme
preference 4
no huntstop
!
!
ephone-dn 29 dual-line
number 8888
conference meetme
preference 1
no huntstop
!
!
ephone-dn 30 dual-line
number 8888
conference meetme
preference 2
no huntstop
!
!
ephone-dn 31 dual-line
number 8888
conference meetme
preference 3
no huntstop
!
!
ephone-dn 32 dual-line
number 8888
conference meetme
preference 4
!
!
ephone-dn 33
number 6033
!
!
ephone-dn 34
number 6034
!
!
ephone-dn 35
number 6035
!
!
ephone-dn 36
number 6036
!
!
ephone-dn 37
number 6037
!
!
ephone-dn 38
number 6038
!
!
ephone-dn 39
number 6039
!
!
ephone-dn 40
number 6040
!
!
ephone-dn 41 dual-line
number 6666
conference meetme
preference 5
no huntstop
!
!
ephone-dn 42 dual-line
number 6666
conference meetme
preference 6
no huntstop
!
!
ephone-dn 43 dual-line
number 6666
conference meetme
preference 7
no huntstop
!
!
ephone-dn 44 dual-line
number 6666
conference meetme
preference 8
no huntstop
!
!
ephone-dn 45 dual-line
number 6666
conference meetme
preference 9
no huntstop
!
!
ephone-dn 46 dual-line
number 6666
conference meetme
preference 10
no huntstop
!
!
ephone-dn 47 dual-line
number 6666
conference meetme
preference 10
no huntstop
!
!
ephone-dn 48 dual-line
number 6666
conference meetme
preference 10
!
!
ephone-dn 51 dual-line
number A0001
name conference
conference ad-hoc
preference 1
no huntstop
!
!
ephone-dn 52 dual-line
number A0001
name conference
conference ad-hoc
preference 2
no huntstop
!
!
ephone-dn 53 dual-line
number A0001
name conference
conference ad-hoc
preference 3
no huntstop
!
!
ephone-dn 54 dual-line
number A0001
name conference
conference ad-hoc
preference 4
!
!
ephone 1
ephone-template 1
mac-address C863.B965.2401
type anl
button 1:1
!
!
!
ephone 2
ephone-template 1
mac-address 0016.C8BE.A04A
type 7920
!
!
!
ephone 3
ephone-template 1
mac-address C863.B965.2400
type anl
button 1:2
!
!
!
ephone 4
no multicast-moh
ephone-template 1
mac-address 0017.952B.7F5C
type 7912
button 1:4
!
!
!
ephone 5
ephone-template 1
ephone 6
no multicast-moh
ephone-template 1
mac-address 0017.594F.1468
type 7961GE
button 1:6
!
!
!
ephone 11
ephone-template 1
mac-address 0016.C8AA.C48C
button 1:10 2:15 3:16 4:17
button 5:18 6:19 7:20 8:21
button 9:22 10:23 11:24 12:33
button 13:34 14:35 15:36 16:37
button 17:38 18:39 19:40
!
!
line con 0
line aux 0
line vty 0 4
login
!
!
end
interface GigabitEthernet0/0
ip address 10.3.111.100 255.255.0.0
duplex auto
speed auto
!
interface GigabitEthernet0/1.1
encapsulation dot1Q 100
ip address 192.168.1.10 255.255.255.0
!
interface GigabitEthernet0/1.2
encapsulation dot1Q 200
ip address 192.168.2.10 255.255.255.0
!
interface GigabitEthernet0/1.3
encapsulation dot1Q 10
ip address 10.15.14.10 255.255.255.0
!
interface GigabitEthernet0/1.4
encapsulation dot1Q 20
ip address 10.15.15.10 255.255.255.0
!
ip route 10.0.0.0 255.0.0.0 10.3.0.1
ip route 192.168.0.0 255.0.0.0 10.3.0.1
!
!
ip http server
!
!
!
!
control-plane
!
sccp local GigabitEthernet0/0
sccp ccm 10.15.15.1 identifier 1 version 4.1
!
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 101 register confer1
associate profile 103 register xcode1
!
!
dspfarm profile 103 transcode
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 6
associate application SCCP
!
dspfarm profile 101 conference
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 5
associate application SCCP
!
!
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
session-timeout 300
exec-timeout 0 0
password
no login
!
scheduler allocate 20000 1000
!
end
Where to Go Next
Controlling Use of the Conference Soft Key
To block the functioning of the conference (Confrn) soft key without removing the key display, create
and apply an ephone template that contains the features blocked command. For more information, see
“Creating Templates” on page 881.
To remove the conference (Confrn) soft key from one or more phones, create and apply an ephone
template that contains the appropriate softkeys command. For more information, see “Customizing Soft
Keys” on page 829.
Additional References
The following sections provide references related to conferencing.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS voice configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Voice Command Reference
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Technical Support & Documentation http://www.cisco.com/techsupport
website contains thousands of pages of searchable
technical content, including links to products,
technologies, solutions, technical tips, and tools.
Registered Cisco.com users can log in from this page to
access even more content.
Note Table 33 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the directory services support available in Cisco Unified Communications
Manager Express (Cisco Unified CME).
Contents
• Information About Directory Services, page 689
• How to Configure Directory Services, page 691
• Configuration Examples for Directory Services, page 701
• Additional References, page 706
• Feature Information for Directory Services, page 707
Local Directory
Cisco Unified CME automatically creates a local phone directory containing the telephone numbers that
are assigned in the directory number configuration of the phone. You can make additional entries to the
local directory in telephony services configuration mode. Additional entries can be nonlocal numbers
such as telephone numbers on other Cisco Unified CME systems used by your company.
When a phone user selects the Directories > Local Directory menu, the phone displays a search page
from Cisco Unified CME. After a user enters the search information, the phone sends the information to
Cisco Unified CME, which searches for the requested number or name pattern in the directory number
configuration and sends the response back to the phone, which displays the matched results. The phone
can display up to 32 directory entries. If a search results in more than 32 entries, the phone displays an
error message and the user must refine the search criteria to narrow the results.
The order of the names in the directory entries can display with first names first or last names first.
The local directory that is displayed on an IP phone is an XML page that is accessed through HTTP
without password protection. The directory HTTP service can be disabled to suppress the availability of
the local directory.
For configuration information, see the “Configuring Local Directory Service” section on page 691.
External Directory
Cisco Unified IP Phones can support URLs in association with the four programmable feature buttons
on IP phones, including the Directories button. Operation of these services is determined by the Cisco
Unified IP phone capabilities and the content of the referenced URL. Provisioning the directory URL to
select an external directory resource disables the Cisco Unified CME local directory service.
Called-Name Display
When phone agents answer calls for several different departments or people, it is often helpful for them
to see a display of the name, rather than the number, of the called party. For example, if order-entry
agents are servicing three catalogs with individual 800 numbers configured in one overlay ephone-dn
set, they need to know which catalog is being called to give the correct greeting, such as “Thank you for
calling catalog N. May I take your order?” The called-name display feature can display either of the
following types of name:
• Name for a directory number in a local directory
• Name associated with an overlay directory number. Calls to the first directory number in a set of
overlay numbers will display a caller ID. Calls to the remaining directory numbers in the overlay set
will display the name associated with the directory number.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. directory {first-name-first | last-name-first}
5. no service local-directory
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# directory
last-name-first
Step 5 no service local-directory Disables local directory service on IP phones.
Example:
Router(config-telephony)# no service
local-directory
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Prerequisites
• Cisco CME 3.0 or a later version.
• Directory number for which you are defining a directory entry must already have a number assigned
by using the number (ephone- dn) command. For configuration information, see “SCCP: Creating
Directory Numbers” on page 158.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. name name
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
Example:
Router(config)# ephone-dn 55
Step 4 name name Associates a name with this directory number. This name is
used for caller-ID displays and in the local directory
listings.
Example:
Router(config-ephone-dn)# name Smith, John • Must follow the name order that is specified with the
directory command.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Restrictions
If the directory entry being configured is to be used for called-name display, the number being
configured must contain at least one wildcard character.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. directory entry {entry-tag number name name | clear}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 directory entry {entry-tag number name Creates a telephone directory entry that is displayed on an IP
name | clear} phone. Entries appear in the order in which they are entered.
• entry-tag—Unique sequence number that identifies this
Example: directory entry during all configuration tasks. Range is
Router(config-telephony)# directory entry 1 to 100.
1 5550111 name Sales
• If this name is to be used for called-name display, the number
associated with the names must contain at least one wildcard
character.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
Prerequisites
To use a Cisco Unified Communications Manager directory as an external directory source for Cisco
Unified CME phones, the Cisco Unified Communications Manager must be made aware of the phones.
You must list the MAC addresses of the Cisco Unified CME phones in the Cisco Unified
Communications Manager and reset the phones from the Cisco Unified Communications Manager. It is
not necessary for you to assign ephone-dns to the phones or for the phones to register with Cisco Unified
Communications Manager.
Restrictions
Provisioning of the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony -service
4. url {directory | service} url
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 url {directory | service} url Associates a URL with the programmable Directories and
Services feature buttons on supported Cisco Unified IP
phones in Cisco Unified CME.
Example:
Router(config-telephony)# url directory • Provisioning the directory URL to select an external
http://10.0.0.11/localdirectory directory resource disables the Cisco Unified CME
local directory service.
• Operation of these services is determined by the
Cisco Unified IP phone capabilities and the content of
the specified URL.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end
Prerequisites
• For directory numbers other than overlaid directory numbers—To display a name in the called-name
display, the name to be displayed must be defined in the local directory. See the “SCCP: Adding an
Entry to a Local Directory” section on page 693.
• For overlaid directory numbers—To display a name in the called-name display for a directory
number that is in a set of overlaid directory numbers, the name to be displayed must be defined. See
the “SCCP: Defining a Name for a Directory Number” section on page 692
Restrictions
• The service dnis overlay command can only be used to configure overlaid ephone-dns.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service dnis dir-lookup
5. service dnis overlay
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)#
Example:
Router(config-telephony)# end
telephony-service
service dnis overlay
Step 2 Use the show telephony-service directory-entry command to display current directory entries.
Router# show telephony-service directory-entry
Step 3 Use the show telephony-service ephone-dn command to verify that you have used at least one wildcard
(period or .) in the ephone-dn primary or secondary number or to verify that you have entered a name
for the number.
Router# show telephony-service ephone-dn
ephone-dn 2
number 5002 secondary 200.
name catalogN
huntstop
call-forward noan 5001 timeout 8
Step 4 Use the show ephone overlay command to verify the contents of overlaid ephone-dn sets.
Router# show ephone overlay
Prerequisites
• Cisco CME 3.4 or a later version.
• Directory number for which you are defining a name must already have a number assigned by using
the number (voice register dn) command. For configuration information, see “SIP: Creating
Directory Numbers” on page 162.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. name name
5. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-register-dn)# name John Smith
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end
Prerequisites
Cisco CME 3.4 or a later version.
Restrictions
• Provisioning of the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.
• Supported only on Cisco Unified IP Phone 7960s and 7960Gs and Cisco Unified IP Phone 7940s
and 7940Gs.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. url {directory | service} url
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 url {directory | service} url Associates a URL with the programmable Directories and
Services feature buttons on supported Cisco Unified Ip
phones in Cisco Unified CME.
Example:
Router(config-register-global)# url directory • Provisioning the directory URL to select an external
http://10.0.0.11/localdirectory directory resource disables the Cisco Unified CME
Router(config-register-global)# url service
local directory service.
http://10.0.0.4/CCMUser/123456/urltest.html
• Operation of these services is determined by the
Cisco Unified IP phone capabilities and the content of
the specified URL.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
SUMMARY STEPS
1. show running-config
2. show telephony-service
3. show telephony-service directory-entry
DETAILED STEPS
.
.
timeout busy 10
timeout ringing 100
caller-id name-only: enable
system message XYZ Company
web admin system name admin1 password admin1
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp Jan 01 2002 00:00:00
transfer-system full-consult
multicast moh 239.12.20.123 port 2000
fxo hook-flash
local directory service: enabled.
Local Directory
The following example defines the naming order for the local directory on IP phones served by the
Cisco Unified CME router:
telephony-service
directory last-name-first
The following example disables the local directory on IP phones served by the Cisco Unified CME
router:
telephony-service
no service local-directory
Called-Name Display
This section contains the following examples:
• First Ephone-dn in the Overlay Set: Example, page 702
• Directory Name for an Overlaid Ephone-dn Set: Example, page 702
• Directory Name for a Hunt Group with Overlaid Ephone-dns: Example, page 703
• Directory Name for Non-Overlaid Ephone-dns: Example, page 704
• Ephone-dn Name for Overlaid Ephone-dns: Example, page 705
ephone-dn 1
number 18005550100
ephone-dn 2
name department1
number 18005550101
ephone-dn 3
name department2
number 18005550102
ephone 1
button 1o1,2,3
ephone 2
button 1o1,2,3
ephone 3
button 1o1,2,3
The default display for all three phones is the number of the first ephone-dn listed in the overlay set
(18005550100). A call is made to the first ephone-dn (18005550100), and the caller ID (for example,
4085550123) is displayed on all three phones. The user for phone 1 answers the call. The caller ID
(4085550123) remains displayed on phone 1, and the displays on phone 2 and phone 3 return to the
default display (18005550100). A call to the next ephone-dn is made. The default display on phone 2
and phone 3 is replaced with the called ephone-dn’s name (18005550101).
ephone-dn 1
number 5500 secondary 555000.
ephone-dn 2
number 5501 secondary 555001.
ephone-dn 3
number 5502 secondary 555002.
ephone 1
button 1o1,2,3
mac-address 1111.1111.1111
ephone 2
button 1o1,2,3
mac-address 2222.2222.2222
ephone 3
button 1o1,2,3
mac-address 3333.3333.3333
For more information about making directory entries, see the “Local Directory” section on page 690. For
more information about overlaid ephone-dns, see “Configuring Call-Coverage Features” on page 563.
ephone-dn 1
number 1001
ephone-dn 2
number 1002
ephone-dn 3
number 1003
ephone-dn 4
number 104
ephone 1
button 1o1,2
button 2o3,4
mac-address 1111.1111.1111
ephone 2
button 1o1,2
button 2o3,4
mac-address 2222.2222.2222
ephone-hunt 1 peer
pilot number 5100 secondary 555....
list 1001, 1002, 1003, 1004
final number 5556000
hops 5
preference 1
timeout 20
no-reg
For more information about hunt-group behavior, see “Configuring Call-Coverage Features” on
page 563. Note that wildcards are used only in secondary numbers and cannot be used with primary
numbers. For more information about making directory entries, see the “Local Directory” section on
page 690. For more information about overlaid ephone-dns, see “Configuring Call-Coverage Features”
on page 563.
ephone-dn 1
number 1001 secondary 555000.
ephone-dn 2
number 1002 secondary 555001.
ephone 1
button 1:1
button 2:2
mac-address 1111.1111.1111
ephone 2
button 1:1
button 2:2
mac-address 2222.2222.2222
ephone 3
button 1:1
button 2:2
mac-address 3333.3333.3333
For more information about making directory entries, see the “Local Directory” section on page 690.
ephone-dn 1
number 18005550000
ephone-dn 2
name catalog1
number 18005550001
ephone-dn 3
name catalog2
number 18005550002
ephone-dn 4
name catalog3
number 18005550003
ephone 1
button 1o1,2,3,4
ephone 2
button 1o1,2,3,4
ephone 3
button 1o1,2,3,4
For more information about overlaid ephone-dns, see “Configuring Call-Coverage Features” on
page 563.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 34 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the do-not-disturb feature in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Do Not Disturb, page 709
• How to Configure Do Not Disturb, page 710
• Configuration Examples for Do Not Disturb, page 714
• Where to Go Next, page 714
• Additional References, page 715
• Feature Information for Do Not Disturb, page 716
Do Not Disturb
The Do Not Disturb (DND) feature allows you to set your phone to forward calls without ringing the
phone. Enable DND service using the DND soft key on Cisco Unified IP phones that support soft keys.
When DND is enabled, incoming calls do not ring on the phone, but they do provide visual alerting and
call information and can be answered if desired. When a local IP phone calls another local IP phone that
is in the DND state, the message “Ring out DND” is displayed on the calling phone indicating that the
target phone is in the DND state.
Pressing the DND soft key during an incoming call forwards the call to the call-forward no-answer
destination if call-forward no-answer is enabled. If call-forward no-answer is not enabled, pressing the
DND soft key disables the ringer and visual alerting.
You can use the DND soft key to switch on or off the DND functionality in all call states except
connected. That is, you can enable or disable DND when an incoming call is ringing or when you are
not connected to a call. You cannot enable or disable DND when you are connected to an incoming call.
In Cisco CME 3.2.1 and later versions, DND can be blocked from phones with the feature-ring function.
A feature ring is a triple-pulse ring, a type of ring cadence in addition to internal call and external call
ring cadences. For example, an internal call in the United States rings for 2 seconds on and 4 seconds
off (single-pulse ring), and an external call rings for 0.4 seconds on, 0.2 seconds off, 0.4 seconds on, and
0.2 seconds off (double-pulse ring).
The triple-pulse ring is used as an audio identifier for phone users. For example, each salesperson in a
sales department could have an IP phone with a button sharing the same set of ephone-dns with the sales
staff and another button for their private line for preferred customers. To help a salesperson identify an
incoming call to his or her private line, the private line can be configured with the feature-ring function.
You can disable the DND function on feature-ring lines. In the preceding example, salespeople could
activate DND on their phones and still hear calls to their private lines.
Prerequisites
• Cisco Unified 3.2.1 or a later version.
• Call-forwarding no-answer must be set for a phone to use DND to forward calls. No other
configuration is necessary for basic DND.
Restrictions
• Phone users cannot enable DND for a shared line in a hunt group. The soft key displays in the idle
and ringing states but does not enable DND for shared lines in hunt groups.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. no dnd feature-ring
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the ephone to be configured.
Router(config)# ephone 10
Step 4 no dnd feature-ring Allows phone buttons configured with the feature-ring
option to ring when their IP phones are in do-not-disturb
(DND) mode.
Example:
Router(config-ephone)# no dnd feature-ring
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Note You can enable the Do-Not-Disturb (DND) soft key on one or more SIP phones by using the
dnd-control command in voice register template configuration mode. For information about
configuring templates, see “Creating Templates” on page 881.
Prerequisites
Cisco CME 3.4 or a later version.
Restrictions
If the DND soft key is disabled by a user on the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G,
7961GE, 7970G, or 7971GE, it does not display after the phone is reset or restarted. DND must be
enabled both in Cisco Unified CME and by using the DND soft key on the phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. dnd
5. exit
6. voice register dn dn-tag
7. call-forward b2bua noan directory- number timeout seconds
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
parameters for SIP phone to be configured.
Example:
Router(config)# voice register pool 1
ephone-dn 2
number 1002
ephone-dn 10
number 1110
preference 0
no huntstop
ephone-dn 11
number 1111
preference 1
ephone 1
button 1f1
button 2o10,11
no dnd feature-ring
ephone 2
button 1f2
button 2o10,11
no dnd feature-ring
Where to Go Next
Agent Status Control for Ephone Hunt Groups and Cisco Unified CME B-ACD
Ephone hunt group agents can control their ready/not-ready status (their ability to receive calls) using
the DND function or the HLog function of their phones. When they use the DND soft key, they do not
receive calls on any extension on their phones. When they use the HLog soft key, they do not receive
calls on hunt group extensions, but they do receive calls on other extensions. For more information on
agent status control and the HLog function, see “Configuring Call-Coverage Features” on page 563.
Call Forwarding
To use the DND soft key to forward calls, enable call-forwarding no-answer. See “Configuring Call
Transfer and Forwarding” on page 499.
Soft-Key Display
You can remove or change the position of the DND soft key. See “Customizing Soft Keys” on page 829.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 35 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
Contents
• Information About Extension Mobility, page 717
• How to Enable Extension Mobility, page 718
• Configuration Examples for Extension Mobility, page 725
• Where to Go Next, page 726
• Additional References, page 727
• Feature Information for Extension Mobility, page 728
Extension Mobility
Extension mobility in Cisco Unified CME 4.2 and later versions provides the benefit of phone mobility
for end users.
A user login service allows phone users to temporarily access a physical phone other than their own
phone and utilize their personal settings, such as directory number, speed-dial lists, and services, as if
the phone is their own desk phone. The phone user can make and receive calls on that phone using the
same personal directory number as is on their own desk phone.
Each Cisco Unified IP phone that is enabled for extension mobility is configured with a logout profile.
This profile determines the default appearance of a phone that is enabled for extension mobility when
there is no phone user logged into that phone. Minimally, the logout profile allows calls to emergency
services such as 911. A single logout profile can be applied to multiple phones.
After a Cisco Unified IP phone that is enabled for extension mobility boots up, the Services button on
the phone is configured with a login service URL hosted by Cisco Unified CME that points to the
extension mobility Login page.
A phone user logs in to a Cisco Unified IP phone that is enabled for extension mobility by pressing the
Services button or a Unified CCX agent can log in using a Unified CCX Cisco Agent Desktop. User
authentication and authorization is performed by Cisco Unified CME. If the login is successful,
Cisco Unified CME retrieves the appropriate user profile, based on user name and password match, and
replaces the phone’s logout profile with the user profile.
After the phone user is logged in, the service URL points to a logout URL hosted by Cisco Unified CME
to provide a logout prompt on the phone. Logging into a different device automatically closes the first
session and start a new session on the new device. When a phone user is not logged in to any phone,
incoming calls to the phone user’s directory number are sent to the phone user’s voice mailbox.
For button appearance, extension mobility associates directory numbers, then speed-dial numbers in the
logout profile or user profile to phone buttons in a sequence. If the profile contains more numbers than
there are buttons on the physical phone to which the profile is downloaded, the remaining numbers in
the profile are ignored.
For configuration information, see the “How to Enable Extension Mobility” section on page 718.
Prerequisites
• Cisco Unified CME 4.2 or a later version.
Restrictions
• Extension mobility on remote Cisco Unified CME routers is not supported; a phone user can log into
any local Cisco Unified IP phone only.
Prerequisites
• All directory numbers to be included in a logout profile or a voice-user profile must be already
configured in Cisco Unified CME. For configuration information, see “Configuring Phones to Make
Basic Calls” on page 147.
Restrictions
• For button appearance, extension mobility associates directory numbers, then speed-dial definitions
in the logout profile or voice-user profile to phone buttons in a sequence beginning with numbers,
followed by speed dials. If the profile contains more directory numbers and speed-dial numbers than
there are buttons on the physical phone to which the profile is downloaded, not all numbers will be
downloaded to buttons.
• The first number to be configured for line appearance cannot be a monitored directory number.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice logout-profile tag
4. username username password password
5. number number type type
6. speed-dial speed-tag number [label label] [blf]
7. pin pin
8. end
DETAILED STEPS
Example:
Router# configure terminal
Prerequisites
• Logout profile to be assigned to a phone must be configured in Cisco Unified CME.
Restrictions
• Extension mobility is not supported on Cisco Unified IP phones without phone screens.
• Extension mobility is not supported for SIP IP phones.
• Extension mobility is not supported for analog devices.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address mac-address
5. type phone-type
6. logout-profile profile-tag
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enables phone configuration mode.
• phone-tag—Unique number that identifies this phone
Example: during configuration tasks. Range is 1 to maximum
Router(config)# ephone 1 number supported phones, where maximum is platform
and version dependent and defined by using the
max-ephone command.
Step 4 mac-address mac-address Associates a physical phone with this ephone configuration.
• mac-address—MAC address of phone, which is found
Example: on a sticker located on the bottom of the phone.
Router(config-ephone)# mac-address
000D.EDAB.3566
Step 5 logout-profile profile-tag Enables Cisco Unified IP phone for extension mobility and
assigns a logout profile to this phone.
Example: • tag—Unique identifier of logout profile to be used
Router(config-ephone)# logout-profile 1 when no phone user is logged in to this phone. This tag
number corresponds to a tag number created when this
logout profile was configured by using the voice
logout-profile command.
Step 6 type phone-type Defines a phone type for the phone being configured.
Example:
Router(config-ephone)# type 7960
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end
Note Templates created using the ephone-template and ephone-dn-template commands can be applied to a
user profile for extension mobility.
Prerequisites
• All directory numbers to be included in a logout profile or voice-user profile must be already
configured in Cisco Unified CME. For configuration information, see “Configuring Phones to Make
Basic Calls” on page 147.
Restrictions
• For button appearance, extension mobility associates directory numbers, then speed-dial definitions
in the logout profile or voice-user profile to phone buttons in a sequence beginning with numbers,
followed by speed dials. If the profile contains more directory numbers and speed-dial numbers than
there are buttons on the physical phone to which the profile is downloaded, not all numbers will be
downloaded to buttons.
• The first number to be configured for line appearance cannot be a monitored directory number.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice user-profile profile-tag
4. name username password password
5. number number type type
6. speed-dial speed-tag number [label label] [blf]
7. pin pin
8. end
DETAILED STEPS
Example:
Router# configure terminal
ephone 2
mac-address 0012.DA8A.C43D
type 7970
logout-profile 1
ephone 3
mac-address 1200.80FC.9B01
type 7911
logout-profile 1
Where to Go Next
• If you created a new or modified an existing logout or voice-user profile, you must restart the phones
to propagate the changes. See “Resetting and Restarting Phones” on page 257.
• If you enabled one or more Cisco Unified IP phones for extension mobility, generate a new
configuration file and restart the phones. See “Generating Configuration Files for Phones” on
page 245.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 36 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the feature access codes support in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Information About Feature Access Codes, page 729
• How to Configure Feature Access Codes, page 731
• Configuration Examples for Feature Access Codes, page 733
• Additional References, page 734
• Feature Information for Feature Access Codes, page 735
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. fac {standard | custom {alias alias-tag custom-fac to existing-fac [extra-digits]} | feature
custom-fac}}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
ephone-hunt hlog-phone *5
The following example shows the output when custom FACs are configured:
Router# show telephony-service fac
FAC: Example
The following example shows how to enable standard FACs for all phones:
Router# telephony-service
Router(config-telephony)# fac standard
fac standard is set!
Router(config-telephony)#
The following example shows how the standard FAC for the Call Forward All feature is changed to a
custom FAC (#45). Then an alias is created to map a second custom fac to #45 plus an extension (1111).
The custom FAC (#44) allows the phone user to press #44 to forward all calls all calls to extension 1111,
without requiring the phone user to dial the extra digits that are the extension number.
Router# telephony-service
Router(config-telephony)# fac custom callfwd all #45
fac callfwd all code has been configured to #45
Router(config-telephony)# fac custom alias 0 #44 to #451111
fac alias0 code has been configurated to #44!
alias0 map code has been configurated to #451111!
The following example shows how to define an alias for the group pickup of group 123. The alias
substitutes the digits #4 for the standard FAC for group pickup (**4) and add s the the group number
(123) to the dial pattern. Using this custom FAC, a phone user can dial #4 to pick up a ringing call in
group 123, instead of dialing the standard FAC **4 plus the group number 123.
Router# telephony-service
Router(config-telephony)# fac custom alias 5 #4 to **4123
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 38 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This module describes how to enable Skinny Client Control Protocol (SCCP) Fax Relay for analog
foreign exchange service (FXS) ports under the control of Cisco Unified CME.
Contents
• Prerequisites for Fax Relay, page 737
• Restrictions for Fax Relay, page 738
• Information About Fax Relay, page 738
• How to Configure Fax Relay, page 740
• Configuration Examples for Fax Relay, page 742
• Additional References, page 743
• Feature Information for Fax Relay, page 744
Note If your voice gateway is a separate router than the Cisco Unified CME router, it must use an IP
voice image of Cisco IOS Release 12.4(11)T or later.
Note For Cisco Unified CME versions before Cisco Unified CME, 4.0(3), there are two manually-controlled
options for setting up facsimiles:
• Fax Gateway Protocol
– Configure the Cisco VG 224, FXS port, or analog telephone adaptor (ATA) to use H.323 or
Session Initiation Protocol (SIP) with a specific fax relay protocol. See the Cisco IOS Fax and
Modem Services over IP Application Guide.
• G.711 Fax Pass-Through with SCCP
– This is the default setup for facsimile on Cisco VG 224 and FXS ports before
Cisco Unified CME 4.0(3). See the Cisco IOS Fax and Modem Services over IP Application
Guide.
VoIP WAN
Cisco gateway FXS Cisco Unified CME Cisco Unified CME FXS Cisco gateway
FXS
FXS V V
PSTN
LAN 1 LAN 2
IP IP
230565
SCCP
For information on configuring gateway-controlled fax relay features, see the “How to Configure Fax
Relay” section on page 740.
Supported Gateways, Modules, and Voice Interface Cards for Fax Relay
Table 39 lists supported gateways, modules, and voice interface cards (VICs).
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode and specifies VoIP
encapsulation.
Example:
Router(config)# voice service voip
Step 4 fax protocol cisco Specifies the Cisco-proprietary Fax Protocol as the fax
protocol for SCCP analog endpoints.
Example: • This command is enabled by default.
Router(config-voi-serv)# fax protocol cisco
Step 5 fax-relay sg3-to-g3 (Optional) Enables the fax stream between two SG3 fax
machines to negotiate down to G3 speeds.
Example:
Router(config-voi-serv)# fax relay sg3-to-g3
Step 6 exit Exits the current configuration mode.
Example:
Router(config-voi-serv)# exit
Troubleshooting Tips
The following commands can help troubleshoot SCCP fax relay features:
• debug voip application stcapp all
• debug voip vtsp all
Note For more information on these and other commands, see the Cisco IOS Voice Command Reference,
Cisco IOS Debug Command Reference, Release 12.4T, Cisco Unified Communications Manager
Express Command Reference, and Cisco IOS Configuration Fundamentals Command Reference,
Release 12.4.
Note For more information on these and other commands, see the Cisco IOS Voice Command Reference,
Cisco IOS Debug Command Reference, Release 12.4T, Cisco Unified Communications Manager
Express Command Reference, and Cisco IOS Configuration Fundamentals Command Reference,
Release 12.4.
ephone-dn 44
number 1234
name fax machine
ephone 33
mac-address 1111.2222.3333
button 1:44
type anl
Additional References
The following sections provide references related to Cisco Fax Relay.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration Cisco Unified Communications Manager Express System
Administrator Guide
Cisco Unified CME command reference Cisco Unified Communications Manager Express Command
Reference
Cisco IOS debugging Cisco IOS Debug Command Reference, Release 12.4T
Cisco IOS voice commands Cisco IOS Voice Command Reference
Cisco IOS voice configuration Cisco IOS Voice Configuration Library
Fax and modem transmission on Cisco Voice over IP Cisco IOS Fax and Modem Service over IP Application Guide
(VoIP) networks
SCCP gateway controlled feature mode call control Feature Mode for SCCP FXS Ports in Cisco IOS
SCCP gateway controlled VMWI VMWI for SCCP FXS Ports in Cisco IOS
SCCP gateway controlled supplementary features SCCP Controlled Analog (FXS) Ports with Supplementary Features
in Cisco IOS Gateways
Platform-specific documentation for the following: http://www.cisco.com/en/US/products/hw/routers/index.html
• Cisco 2800 Series Integrated Services Routers http://www.cisco.com/en/US/products/sw/voicesw/index.html
• Cisco 3800 Series Integrated Services Routers
• Cisco VG 224 Voice Gateway Router
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies.
To receive security and technical information about
your products, you can subscribe to various services,
such as the Product Alert Tool (accessed from Field
Notices), the Cisco Technical Services Newsletter, and
Really Simple Syndication (RSS) Feeds.
Access to most tools on the Cisco Support website
requires a Cisco.com user ID and password.
Note Table 40 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the headset auto-answer feature in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Information About Headset Auto-Answer, page 745
• How to Configure Headset Auto-Answer, page 748
• Configuration Examples for Headset Auto-answer, page 749
• Additional References, page 750
• Feature Information for Headset Auto-Answer, page 751
Most of the time, a line number is the same as the button number on which
it appears.
In this example, line 1 is button 1, line 2 is button 2, and line 3 is button 3.
ephone-dn 21
number 2001
ephone-dn 22
number 2002
ephone-dn 23
number 2003
2001 Li ne 1
ephone 2
2002 Li ne 2
button 1:21 2:22 3:23
2003 Li ne 3 headset auto-answer line 1
headset auto-answer line 2
But not always. In the following case, line 2 is button 3, because
button3 is the second button that has an ephone-dn to be connected
to a phone call. Button 2 is unoccupied and cannot take calls.
ephone-dn 33
number 2889
ephone-dn 34
number 2887
ephone 2
2889 Li ne 1 button 1:33 3:34
headset auto-answer line 1
2887 Li ne 2 headset auto-answer line 2
ephone-dn 21
In the following example, button 2 has three overlay ephone- number 2001
dns (22, 23, and 24). Button 2 is defined as one line because
only one of those ephone-dns can be connected to a call ephone-dn 22
using this button at any one time. number 2002
ephone-dn 23
number 2003
ephone-dn 24
number 2004
ephone-dn 25
number 2005
2001 Li ne 1
2002, 2003, 2004 Li ne 2 ephone 2
2005 Li ne 3 button 1:21 2o22,23,24 3:25
headset auto-answer line 2
headset auto-answer line 3
2004 Li ne 3
headset auto-a nswer line 1
headset auto-a nswer line 2
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. headset auto-answer line line-number
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks. The maximum number of
Router(config)# ephone 25 ephones for a particular Cisco Unified CME system is
version- and platform-specific. For the range of values, see
the CLI help.
Step 4 headset auto-answer line line-number Specifies a line on an ephone that will be answered automatically
when the headset button is depressed.
Example: • line-number—Number of the phone line that should be
Router(config-ephone)# headset automatically answered.
auto-answer line 1
Note Repeat this command to add additional lines.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
ephone 1
headset auto-answer line 1
headset auto-answer line 2
headset auto-answer line 3
headset auto-answer line 4
username "Front Desk"
mac-address 011F.92B0.BE03
speed-dial 1 330 label “Billing”
type 7960 addon 1 7914
no dnd feature-ring
keep-conference
button 1f40 2f41 3f42 4:30
button 5:405 7m20 8m21 9m22
button 10m23 11m24 12m25 13m26
button 14m499 15:1 16m31 17f498
button 18s500
night-service bell
Step 2 Use the show telephony-service ephone command to display only the ephone configuration portion of
the running configuration.
The following example enables headset auto-answer on ephone 17 for line 2 (button 2), which has
overlaid ephone-dns, and line 3 (button 3), which is an overlay rollover line.
ephone 17
button 1:2 2o21,22,23,24,25 3x2
headset auto-answer line 2
headset auto-answer line 3
The following example enables headset auto-answer on ephone 25 for line 2 (button 3) and
line 3 (button 5). In this case, the button numbers do not match the line numbers because buttons 2 and 4
are not used.
ephone 25
button 1:2 3:4 5:6
headset auto-answer line 2
headset auto-answer line 3
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 41 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the intercom features in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Intercom Lines, page 753
• How to Configure Intercom Lines, page 755
• Configuration Examples for Intercom Lines, page 759
• Where to Go Next, page 759
• Additional References, page 760
• Feature Information for Intercom Lines, page 761
sounded when the call is auto-answered to alert the recipient to the incoming call. To respond to the
intercom call and open a two-way voice path, the recipient deactivates the mute function by pressing the
Mute button or, on phones such as the Cisco Unified IP Phone 7910, lifting the handset.
In Cisco CME 3.2.1 and later versions, you can deactivate the speaker-mute function on intercom calls.
For example, if phone user 1 makes an intercom call to phone user 2, both users hear each other on
connection when no-mute is configured. The benefit is that people who receive intercom calls can be
heard without them having to disable the mute function. The disadvantage is that nearby background
sounds and conversations can be heard the moment a person receives an intercom call, regardless of
whether they are ready to take a call or not.
Intercom lines cannot be used in shared-line configurations. If a directory number is configured for
intercom operation, it must be associated with one IP phone only. The intercom attribute causes an IP
phone line to operate as an autodial line for outbound calls and as an autoanswer-with-mute line for
inbound calls. Figure 47 shows an intercom between a receptionist and a manager.
To prevent an unauthorized phone from dialing an intercom line (and creating a situation in which a
phone automatically answers a nonintercom call), you can assign the intercom a directory number that
includes an alphabetic character. No one can dial the alphabetic character from a normal phone, but the
phone at the other end of the intercom can be configured to dial the number that contains the alphabetic
character through the Cisco Unified CME router. For example, the intercom ephone-dns in Figure 47 are
assigned numbers with alphabetic characters so that only the receptionist can call the manager on his or
her intercom line, and no one except the manager can call the receptionist on his or her intercom line.
Note An intercom requires the configuration of two ephone-dns, one each on a separate phone.
ephone-dn 2
1 The receptionist at phone 6 2 Phone 7 beeps once and automatically number 2345
makes an intercom call to answers in speakerphone mode with
phone 7 by pressing button 2. mute activated. The manager hears the ephone-dn 3
receptionist’s voice and deactivates the number 4578
mute function to open a two-way voice
path for a reply. ephone-dn 18
number A5001
name "Intercom"
intercom A5002
IP IP
V ephone-dn 19
number A5002
Phone 6 - Receptionist Phone 7 - Manager name "Intercom"
Button 1 is extension 2345, a Button 1 is extension 4578, a intercom A5001
normal line. normal line.
Button 2 is extension A5001, a Button 2 is extension A5002, a ephone 6
dedicated intercom connection dedicated intercom connection to button 1:2 2:18
to intercom extension intercom extension
88952
Restrictions
• Intercom lines cannot be dual-line.
• If a directory number is configured for intercom operation, it can be associated with only one
Cisco Unified IP phone.
• A separate configuration is required for each phone at both ends of the two-way voice path.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number
5. name name
6. intercom directory-number [barge-in | no-auto-answer] [label label] [no-mute]
7. exit
8. ephone phone-tag
9. button button-number:dn-tag [[button-number:dn-tag] ...]
10. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config-ephone-dn)# intercom A2346 label
Security
Step 7 exit Exits ephone-dn configuration mode.
Example:
Router(config-ephone-dn)# exit
Step 8 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 24
Step 9 button button-number:dn-tag [[button-number:dn-tag] Assigns a button number to the intercom ephone-dn
...] being configured.
• Use the colon separator (:) between the button
Example: number and the intercom ephone-dn tag to
Router(config-ephone)# button 1:1 2:4 3:14 indicate a normal ring for the intercom line.
Step 10 end Exits configuration mode and enters privileged
EXEC mode.
Example:
Router(config)# exit
Prerequisites
Cisco CME 3.4 or a later version.
Restrictions
• If a directory number is configured for intercom operation, it can be associated with only one
Cisco Unified IP phone.
• A separate configuration is required for each phone at each end of the two-way voice path.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. auto-answer
6. exit
7. voice register pool pool-tag
8. id mac address
9. type phone-type
10. number tag dn dn-tag
11. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1
Example:
Router(config-register-pool)# id mac
0009.A3D4.1234
Step 9 type phone-type Defines a phone type for the SIP phone being configured.
Example:
Router(config-register-pool)# type 7960-7940
Step 10 number tag dn dn-tag Associates a directory number with the SIP phone being
configured.
Example:
Router(config-register-pool)# number 1 dn 17
Step 11 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end
ephone-dn 4
number 5222
ephone-dn 18
number 5001
name “intercom”
intercom 5002 barge-in
ephone-dn 19
name “intercom”
number 5002
intercom 5001 barge-in
ephone 4
button 1:2 2:18
ephone 5
button 1:4 2:19
Where to Go Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
Paging
The paging feature sets up a one-way audio path to deliver information to a group of phones at one time.
For more information, see “Configuring Paging” on page 785.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 42 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the loopback call-routing feature in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Information About Loopback Call Routing, page 763
• How to Configure Loopback Call Routing, page 764
• Configuration Examples for Loopback Call Routing, page 768
• Additional References, page 769
• Feature Information for Loopback Call Routing, page 770
(Cisco-proprietary or H.450-based). Control messages that request call transfer or call forwarding are
intercepted at the loopback virtual port and serviced on the local voice gateway. If needed, this
mechanism creates VoIP-to-VoIP call-routing paths.
Loopback call routing may be used for routing H.323 calls to Cisco Unity Express. For information on
configuring Cisco Unity Express, see the Cisco Unity Express documentation.
Note A preferred alternative to loopback call routing was introduced in Cisco CME 3.1. This alternative
blocks H.450-based supplementary service requests by using the following Cisco IOS commands:
no supplementary-service h450.2, no supplementary-service h450.3, and supplementary-service
h450.12. For more information, see “Configuring Call Transfer and Forwarding” on page 499.
Use of loopback-dn configurations within a VoIP network should be restricted to resolving critical
network interoperability service problems that cannot otherwise be solved. Loopback-dn configurations
are intended for use in VoIP network interworking where the alternative would be to make use of
back-to-back-connected physical voice ports. Loopback-dn configurations emulate the effect of a
back-to-back physical voice-port arrangement without the expense of the physical voice-port hardware.
Because digital signal processors (DSPs) are not involved in loopback-dn arrangements, the
configuration does not support interworking or transcoding between calls that use different voice codecs.
In many cases, use of back-to-back physical voice ports that do involve DSPs to resolve VoIP network
interworking issues is preferred, because it introduces fewer restrictions in terms of supported codecs
and call flows.
Loopback call routing requires two extensions (ephone-dns) to be separately configured, each as half of
a loopback-dn pair. Ephone-dns that are defined as a loopback-dn pair can only be used for loopback call
routing. In addition to defining the loopback-dn pair, you must specify preference, huntstop, class of
restriction (COR), and translation rules.
Restrictions
Loopback-dns do not support T.38 fax relay.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode, creates an ephone-dn,
and optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 15 ephone-dn during configuration tasks. Range is
platform- and version-dependent.
Note Ephone-dns used for loopback cannot be dual-line
ephone-dns.
Step 4 number number [secondary number] [no-reg Associates a number with this extension (ephone-dn).
[both | primary]]
• number—String of up to 16 digits that represents a
telephone or extension number to be associated with this
Example: ephone-dn.
Router(config-ephone-dn)# number 2001
• secondary—(Optional) Allows you to associate a second
telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should not
register with the H.323 gatekeeper. The no-reg keyword by
itself indicates that only the secondary number should not
register. The no-reg both keywords indicate that both
numbers should not register, and the no-reg primary
keywords indicate that only the primary number should not
register.
Example:
Router(config-ephone-dn)# end
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 43 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the music on hold features in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Music on Hold, page 772
• Prerequisites for Music on Hold, page 771
• Restrictions for Music on Hold, page 771
• How to Configure Music on Hold, page 774
• Configuration Examples for Music on Hold, page 781
• Additional References, page 782
• Feature Information for Music on Hold, page 783
• Cisco Unified CME 4.0 and later versions support MOH for internal calls only if the multicast moh
command is used to enable the flow of packets to the subnet on which the phones are located.
• Internal extensions that are connected through an analog voice gateway (Cisco VG 224) or through
a WAN (remote extensions) do not hear MOH on internal calls.
• Multicast MOH is not supported on a phone if the phone is configured with the mtp command or
the paging-dn command with the unicast keyword.
Music on Hold
Music on hold (MOH) is an audio stream that is played to PSTN and VoIP G.711 or G.729 callers who
are placed on hold by phones in a Cisco Unified CME system. This audio stream is intended to reassure
callers that they are still connected to their calls.
When the phone receiving MOH is part of a system that uses a G.729 codec, transcoding is required
between G.711 and G.729. The G.711 MOH must be translated to G.729. Note that because of
compression, MOH using G.729 is of significantly lower fidelity than MOH using G.711. For
information about transcoding, see “Configuring Transcoding Resources” on page 303.
If the MOH audio stream is also identified as a multicast source, the Cisco Unified CME router
additionally transmits the stream on the physical IP interfaces of the Cisco Unified CME router that you
specify during configuration, which permits external devices to have access to it.
Certain IP phones do not support IP multicast and, therefore, do not support multicast MOH. You can
disable multicast MOH to individual phones that do not support multicast. Callers hear a repeating tone
when they are placed on hold.
The audio stream that is used for MOH can derive from one of two sources:
• Audio file—A MOH audio stream from an audio file is supplied from an .au or .wav file held in
router flash memory.
• Live feed—A MOH audio stream from a live feed is supplied from a standard line-level audio
connection that is directly connected to the router through an FXO or “ear and mouth” (E&M)
analog voice port.
If both are configured concurrently on the Cisco Unified CME router, the router seeks the live feed first.
If the live feed is found, it displaces the audio file source. If the live feed is not found or fails at any time,
the router falls back to the audio file source that was specified for MOH during configuration.
Configuring MOH for SIP phones is the same as configuring MOH for SCCP phones. For configuration
information, see the “How to Configure Music on Hold” section on page 774.
Prerequisites
• SIP phones require Cisco Unified CME 4.1 or a later version.
• A music file must be in stored in the router’s flash memory. This file should be in G.711 format. The
file can be in .au or .wav file format, but the file format must contain 8-bit 8-kHz data; for example,
ITU-T A-law or mu-law data format.
Restrictions
• If MOH from an audio file and MOH from a live feed are both configured on the Cisco Unified CME
router, the router seeks the live feed first. If a live feed is found, it displaces an audio file source. If
the live feed is not found or fails at any time, the router falls back to the audio file source.
• To change the audio file to a different file, you must remove the first file using the no moh command
before specifying a second file. If you configure a second file without removing the first file, the
MOH mechanism stops working and may require a router reboot to clear the problem.
• The volume level of a MOH file cannot be adjusted through Cisco IOS software, so it cannot be
changed when the file is loaded into the flash memory of the router. To adjust the volume level of a
MOH file, edit the file in an audio editor before downloading the file to router flash memory.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. moh filename
5. multicast moh ip-address port port-number [route ip-address-list]
6. exit
7. ephone phone-tag
8. multicast-moh
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 moh filename Enables music on hold using the specified file.
• filename—Source of the audio stream for MOH.
Example: Note If you specify a file with this command and
Router(config-telephony)# moh minuet.au
later want to use a different file, you must
disable use of the first file with the no moh
command before configuring the second file.
Step 5 multicast moh ip-address port port-number [route Specifies that the MOH audio stream should also be
ip-address-list] multicast as specified.
Note This command is required to use MOH for
Example: internal calls and it must be configured after
Router(config-telephony)# multicast moh 239.10.16.4 MOH is enabled with the moh command.
port 2123 route 10.10.29.17 10.10.29.33
• ip-address—Specifies that this audio stream is to
be used for multicast and also for MOH, and
specifies the destination IP address for multicast.
• port port-number—Media port for multicast.
Range is 2000 to 65535. We recommend
port 2000 because it is already used for normal
RTP media transmissions between IP phones and
the router.
• route—(Optional) Specifies a list of explicit
router interfaces for the IP multicast packets.
• ip-address-list—(Optional) List of up to four
explicit routes for multicast MOH. The default is
that the MOH multicast stream is automatically
output on the interfaces that correspond to the
address that was configured with the ip
source-address command.
Note For MOH on internal calls, packet flow must
be enabled to the subnet on which the phones
are located.
Example:
Router(config-telephony)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 28
Step 8 multicast-moh (Optional) Enables multicast MOH on a phone. This
is the default.
Example: The no form of this command disables MOH for
Router(config-ephone)# no multicast-moh phones that do not support multicast. Callers hear a
repeating tone when they are placed on hold.
Note This command can also be used in an ephone
template that is applied to one or more
phones.
Step 9 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Prerequisites
• SIP phones require Cisco Unified CME 4.1 or a later version.
• VIC2-2FXO, VIC2-4FXO, EM-HDA-3FXS/4FXO, EM-HDA-6FXO, or EM2-HDA-4FXO
• For a live feed from VoIP, VAD must be disabled.
Restrictions
• A foreign exchange station (FXS) port cannot be used for a live feed.
• The signal loop-start live-feed command for FXO ports is supported in Cisco IOS
Release 12.4(11)XJ, 12.4(15)T, and later releases.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice-port port
4. input gain decibels
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice-port port Enters voice-port configuration mode.
Note Port argument is platform-dependent; type ? to
Example: display syntax. For more information, see the
Router(config)# voice-port 1/1/0 Cisco IOS Voice Command Reference.
Step 4 input gain decibels Specifies, in decibels, the amount of gain to be inserted at
the receiver side of the interface. Acceptable values are
integers from –6 to 14.
Example:
Router(config-voice-port)# input gain 0
Step 5 auto-cut-through (E&M ports only) Enables call completion when a PBX
does not provide an M-lead response. MOH requires that
you use this command with E&M ports.
Example:
Router(config-voice-port)# auto-cut-through
Example:
Router(config-voice-port)# exit
Step 11 dial peer voice tag pots Enters dial-peer configuration mode.
Example:
Router(config)# dial peer voice 7777 pots
Step 12 destination-pattern string Specifies either the prefix or the full E.164 telephone
number to be used for a dial peer.
Example:
Router(config-dial-peer)# destination-pattern
7777
Step 13 port port Associates the dial peer with the voice port that was
specified in Step 3.
Example:
Router(config-dial-peer)# port 1/1/0
Step 14 exit Exits dial-peer configuration mode.
Example:
Router(config-dial-peer)# exit
Step 15 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies this
Example: ephone-dn during configuration tasks. Range is
Router(config)# ephone-dn 55 1 to 288.
Example:
Router(config-ephone-dn)# exit
Step 19 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 28
Example:
Router(config-ephone)# end
telephony-service
fxo hook-flash
load 7960-7940 P00307020300
load 7914 S00104000100
max-ephones 100
max-dn 500
ip source-address 10.123.23.231 port 2000
max-redirect 20
timeouts ringing 100
system message XYZ Company
voicemail 7189
max-conferences 8 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
multicast moh 239.15.10.1 port 2000
web admin system name admin1 password admin1
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
fac custom callfwd all **1
fac custom callfwd cancel **2
fac custom pickup local **3
fac custom pickup group *7
fac custom pickup direct **5
fac custom park *8
fac custom dnd **7
fac custom redial #8
fac custom voicemail **9
fac custom ephone-hunt join *3
fac custom ephone-hunt cancel #3
create cnf-files version-stamp Jan 01 2002 00:00:00
Step 2 Use the show telephony-service command to display only the telephony-service configuration
information.
The following example enables MOH and additionally specifies a multicast address for the audio stream:
telephony-service
moh minuet.wav
multicast moh 239.23.4.10 port 2000
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 44 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the paging feature in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Paging, page 785
• How to Configure Paging, page 787
• Configuration Examples for Paging, page 791
• Where to Go Next, page 794
• Additional References, page 794
• Feature Information for Paging, page 795
Audio Paging
A paging number can be defined to relay audio pages to a group of designated phones. When a caller
dials the paging number (ephone-dn), each idle IP phone that has been configured with the paging
number automatically answers using its speakerphone mode. Displays on the phones that answer the
page show the caller ID that has been set using the name command under the paging ephone-dn. When
the caller finishes speaking the message and hangs up, the phones are returned to their idle states.
Audio paging provides a one-way voice path to the phones that have been designated to receive paging.
It does not have a press-to-answer option like the intercom feature. A paging group is created using a
dummy ephone-dn, known as the paging ephone-dn, that can be associated with any number of local IP
phones. The paging ephone-dn can be dialed from anywhere, including on-net.
After you have created two or more simple paging groups, you can unite them into combined paging
groups. By creating combined paging groups, you provide phone users with the flexibility to page a
small local paging group (for example, paging four phones in a store’s jewelry department) or to page a
combined set of several paging groups (for example, by paging a group that consists of both the jewelry
department and the accessories department).
The paging mechanism supports audio distribution using IP multicast, replicated unicast, and a mixture
of both (so that multicast is used where possible, and unicast is used for specific phones that cannot be
reached using multicast).
Figure 48 shows a paging group with two phones.
1 To page all the phones in the shipping IP Any phone dials 4444.
department, a person at any phone dials
the number associated with the paging
ephone-dn for the shipping department.
The paging ephone-dn has a number that
does not appear on any phone (in this
example, extension 4444). Ephone-dn 4
Extension 4444
This is a paging ephone-dn; no physical phone
instrument is associated with this number.
2 A one-way voice connection is automatically 4444
made with all idle ephones that are
configured with paging ephone-dn 4. In this
example, that is phone 1 and phone 2. Both
phones answer the call in speakerphone V
mode. The voice of the calling party is heard
through the speaker, and the phone displays
the caller ID (name) of paging ephone-dn 4 Phone 1
("Paging Shipping"). Button 1 is extension 2121, a
IP
normal line.
This phone has a paging-dn to
receive pages.
ephone-dn 4
number 4444 Phone 2
name Paging Shipping Button 1 is extension 2222, a normal line.
IP This phone has a paging-dn to receive
paging ip 239.0.1.20 port 2000
pages.
ephone-dn 21
number 2121
ephone 1
mac-address 3662.0234.6ae2
button 1:21
paging-dn 4
ephone 2
88953
mac-address 9387.6738.2873
button 1:22
paging-dn 4
Restrictions
IP phones do not support multicast at 224.x.x.x addresses.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn paging-dn-tag
4. number number
5. name name
6. paging [ip multicast-address port udp-port-number]
7. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn paging-dn-tag Enters ephone-dn configuration mode.
• paging-dn-tag—A unique sequence number that identifies this
Example: paging ephone-dn during all configuration tasks. This is the
Router(config)# ephone-dn 42 ephone-dn that is dialed to initiate a page. This ephone-dn is not
associated with a physical phone. Range is 1 to 288.
Note Do not use the dual-line keyword with this command.
Paging ephone-dns cannot be dual-line.
Example:
Router(config-telephony)# end
Prerequisites
Simple paging groups must be configured. See the “Configuring a Simple Paging Group” section on
page 787.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn paging-dn-tag
4. number number
5. name name
6. paging group paging-dn-tag,paging-dn-tag[[,paging-dn-tag]...]
7. exit
8. ephone phone-tag
9. paging-dn paging-dn-tag {multicast | unicast}
10. exit
11. Repeat Step 8 to Step 10 to add additional IP phones to the paging group.
12. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn paging-dn-tag Enters ephone-dn configuration mode to create a paging number for
a combined paging group.
Example: • paging-dn-tag—A unique sequence number that identifies this
Router(config)# ephone-dn 42 paging ephone-dn during all configuration tasks. This is the
ephone-dn that is dialed to initiate a page. This ephone-dn is not
associated with a physical phone. Range is 1 to 288.
Note Do not use the dual-line keyword with this command.
Paging ephone-dns cannot be dual-line.
Step 4 number number Defines an extension number associated with the combined group
paging ephone-dn. This is the number that people call to initiate a
page to the combined group.
Example:
Router(config-ephone-dn)# number 3556
Step 5 name name (Optional) Assigns to the combined group paging number a name to
appear in caller-ID displays and directories.
Example:
Router(config-ephone-dn)# name paging4
Example:
Router(config-ephone-dn)# exit
Step 8 ephone phone-tag Enters ephone configuration mode to add IP phones to the paging
group.
Example: • phone-tag—Unique sequence number of a phone to receive
Router(config)# ephone 2 audio pages when the paging ephone-dn is called.
Step 9 paging-dn paging-dn-tag {multicast | Associates this ephone with an ephone-dn tag that is used for a
unicast} paging ephone-dn (the number that people call to deliver a page).
Note that the paging ephone-dn tag is not associated with a line
Example: button on this ephone.
Router(config-ephone)# paging-dn 42 The paging mechanism supports audio distribution using IP
multicast
multicast, replicated unicast, and a mixture of both (so that
multicast is used where possible and unicast is allowed to specific
phones that cannot be reached through multicast).
• paging-dn-tag—Unique sequence number for a paging
ephone-dn.
• multicast—(Optional) Multicast paging for groups. By default,
paging is transmitted to the Cisco Unified IP phone using
multicast.
• unicast—(Optional) Unicast paging for a single
Cisco Unified IP phone. This keyword indicates that the
Cisco Unified IP phone is not capable of receiving paging
through multicast and requests that the phone receive paging
through a unicast transmission directed to the individual phone.
Note The number of phones supported through unicast is limited
to a maximum of ten phones.
Step 10 exit Exits ephone configuration mode.
Example:
Router(config-ephone)# exit
Example:
Router(config-telephony)# end
Verifying Paging
Step 1 Use the show running-config command to display the running configuration. Paging ephone-dns are
listed in the ephone-dn portion of the output. Phones that belong to paging groups are listed in the ephone
part of the output.
Router# show running-config
ephone-dn 48
number 136
name PagingCashiers
paging ip 239.1.1.10 port 2000
ephone 2
headset auto-answer line 1
headset auto-answer line 4
ephone-template 1
username "FrontCashier"
mac-address 011F.2A0.A490
paging-dn 48
type 7960
no dnd feature-ring
no auto-line
button 1f43 2f44 3f45 4:31
Step 2 Use the show telephony-service ephone-dn and show telephony-service ephone commands to display
only the configuration information for ephone-dns and ephones.
ephone 4
mac-address 0030.94c3.8724
button 1:1 2:2
paging-dn 22 multicast
In this example, paging calls to 2000 are multicast to Cisco Unified IP phones 1 and 2, and paging calls
to 2001 go to Cisco Unified IP phones 3 and 4. Note that the paging ephone-dns (20 and 21) are not
assigned to any phone buttons.
ephone-dn 20
number 2000
paging ip 239.0.1.20 port 2000
ephone-dn 21
number 2001
paging ip 239.0.1.21 port 2000
ephone 1
mac-address 3662.024.6ae2
button 1:1
paging-dn 20
ephone 2
mac-address 9387.678.2873
button 1:2
paging-dn 20
ephone 3
mac-address 0478.2a78.8640
button 1:3
paging-dn 21
ephone 4
mac-address 4398.b694.456
button 1:4
paging-dn 21
ephone-dn 20
number 2000
paging ip 239.0.1.20 port 2000
ephone-dn 21
number 2001
paging ip 239.0.1.21 port 2000
ephone-dn 22
number 2002
paging ip 239.0.2.22 port 2000
paging group 20,21
ephone-dn 6
number 1103
name user3
ephone-dn 7
number 1104
name user4
ephone-dn 8
number 1105
name user5
ephone-dn 9
number 1199
ephone-dn 10
number 1198
ephone 1
mac-address 1234.8903.2941
button 1:6
paging-dn 20
ephone 2
mac-address CFBA.321B.96FA
button 1:7
paging-dn 20
ephone 3
mac-address CFBB.3232.9611
button 1:8
paging-dn 21
ephone 4
mac-address 3928.3012.EE89
button 1:9
paging-dn 21
ephone 5
mac-address BB93.9345.0031
button 1:10
paging-dn 22
Where to Go Next
Intercom
The intercom feature is similar to paging because it allows a phone user to deliver an audio message to
a phone without the called party having to answer. The intercom feature is different than paging because
the audio path between the caller and the called party is a dedicated audio path and because the called
party can respond to the caller. See “Configuring Intercom Lines” on page 753.
Speed Dial
Phone users who make frequent pages may want to include the paging ephone-dn numbers in their list
of speed-dial numbers. See “Configuring Speed Dial” on page 847.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 45 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This module describes presence support in a Cisco Unified Communications Manager Express
(Cisco Unified CME) system.
Contents
• Prerequisites for Presence Service, page 797
• Restrictions for Presence Service, page 798
• Information About Presence Service, page 798
• How to Configure Presence Service, page 799
• Configuration Examples for Presence, page 813
• Additional References, page 817
• Feature Information for Presence Service, page 818
Presence Service
A presence service, as defined in RFC 2778 and RFC 2779, is a system for finding, retrieving, and
distributing presence information from a source, called a presence entity (presentity), to an interested
party called a watcher. When you configure presence in a Cisco Unified CME system with a SIP WAN
connection, a phone user, or watcher, can monitor the real-time status of another user at a directory
number, the presentity. Presence enables the calling party to know before dialing whether the called
party is available. For example, a directory application may show that a user is busy, saving the caller
the time and inconvenience of not being able to reach someone.
Presence uses SIP SUBSCRIBE and NOTIFY methods to allow users and applications to subscribe to
changes in the line status of phones in a Cisco Unified CME system. Phones act as watchers and a
presentity is identified by a directory number on a phone. Watchers initiate presence requests
(SUBSCRIBE messages) to obtain the line status of a presentity. Cisco Unified CME responds with the
presentity’s status. Each time a status changes for a presentity, all watchers of this presentity are sent a
notification message. SIP phones and trunks use SIP messages; SCCP phones use presence primitives in
SCCP messages.
Presence supports Busy Lamp Field (BLF) notification features for speed-dial buttons and directory call
lists for missed calls, placed calls, and received calls. SIP and SCCP phones that support the BLF
speed-dial and BLF call-list features can subscribe to status change notification for internal and external
directory numbers.
Figure 49 shows a Cisco Unified CME system supporting BLF notification for internal and external
directory numbers. If the watcher and the presentity are not both internal to the Cisco Unified CME
router, the subscribe message is handled by a presence proxy server.
SIP
Subscribe
Notify
V
Subscribe
Notify
155790
IP IP IP IP
IP IP IP IP
The following line states display through BLF indicators on the phone:
• Line is idle—Displays when this line is not being used.
• Line is in-use—Displays when the line is in the ringing state and when a user is on the line, whether
or not this line can accept a new call.
• BLF indicator unknown—Phone is unregistered or this line is not allowed to be watched.
Cisco Unified CME acts as a presence agent for internal lines (both SIP and SCCP) and as a presence
server for external watchers connected through a SIP trunk, providing the following functionality:
• Processes SUBSCRIBE requests from internal lines to internal lines. Notifies internal subscribers
of any status change.
• Processes incoming SUBSCRIBE requests from a SIP trunk for internal SCCP and SIP lines.
Notifies external subscribers of any status change.
• Sends SUBSCRIBE requests to external presentities on behalf of internal lines. Relays status
responses to internal lines.
Presence subscription requests from SIP trunks can be authenticated and authorized. Local subscription
requests cannot be authenticated.
For configuration information, see the “How to Configure Presence Service” section on page 799.
Restrictions
• A presentity can be identified by a directory number only.
• BLF monitoring indicates the line status only.
• Instant Messaging is not supported.
SUMMARY STEPS
1. enable
2. configure terminal
3. sip-ua
4. presence enable
5. exit
6. presence
7. max-subscription number
8. presence call-list
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 sip-ua Enters SIP user-agent configuration mode to configure the
user agent.
Example:
Router(config)# sip-ua
Example:
Router(config-sip-ua)# presence enable
Step 5 exit Exits SIP user-agent configuration mode.
Example:
Router(config-sip-ua)# exit
Step 6 presence Enables presence service and enters presence configuration
mode.
Example:
Router(config)# presence
Step 7 presence call-list Globally enables BLF monitoring for directory numbers in
call lists and directories on all locally registered phones.
Example: • Only directory numbers that you enable for watching
Router(config-presence)# presence call-list with the allow watch command display BLF status
indicators.
• This command enables the BLF call-list feature
globally. To enable the feature for a specific phone, see
the “Enabling a SCCP Phone to Monitor BLF Status for
Speed-Dials and Call Lists” section on page 803.
Step 8 max-subscription number (Optional) Sets the maximum number of concurrent watch
sessions that are allowed.
Example: • number—Maximum watch sessions. Range: 100 to the
Router(config-presence)# max-subscription 128 maximum number of directory numbers supported on
the router platform. Type ? to display range.
Default: 100.
Step 9 end Exits to privileged EXEC mode.
Example:
Router(config-presence)# end
Restrictions
• A presentity is identified by a directory number only.
• BLF monitoring indicates the line status only.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
or
voice register dn dn-tag
4. number number
5. allow watch
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters the configuration mode to define a directory number
or for an IP phone, intercom line, voice port, or a
voice register dn dn-tag message-waiting indicator (MWI).
• dn-tag—Identifies a particular directory number during
Example: configuration tasks. Range is 1 to the maximum number
Router(config)# ephone-dn 1 of directory numbers allowed on the router platform, or
or the maximum defined by the max-dn command. Type
? to display range.
Router(config)# voice register dn 1
Step 4 number number Associates a phone number with a directory number to be
assigned to an IP phone in Cisco Unified CME.
Example: • number—String of up to 16 characters that represents
Router(config-ephone-dn)# number 3001 an E.164 telephone number.
or
Router(config-register-dn)# number 3001
Example:
Router(config-ephone-dn)# end
or
Router(config-register-dn)# end
Enabling a SCCP Phone to Monitor BLF Status for Speed-Dials and Call Lists
A watcher can monitor the status of lines associated with internal and external directory numbers
(presentities) through the BLF speed-dial and BLF call-list presence features. To enable the BLF
notification features on an IP phone using SCCP, perform the following steps.
Prerequisites
• Presence must be enabled on the Cisco Unified CME router. See the “Enabling Presence for Internal
Lines” section on page 800.
• A directory number must be enabled as a presentity with the allow watch command to provide BLF
status notification. See the “Enabling a Directory Number to be Watched” section on page 801.
Restrictions
BLF Call-List
• Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
BLF Speed-Dial
• Supported only on Cisco Unified IP Phone 7914, 7931, 7940, 7941G, 7941GE, 7960, 7961G,
7961GE, 7970G, and 7971GE.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode to set phone-specific
parameters for a SIP phone.
Example: • phone-tag—Unique sequence number of the phone to
Router(config)# ephone 1 be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-ephones command.
Step 4 button button-number{separator}dn-tag Associates a button number and line characteristics with a
[,dn-tag...] directory number on the phone.
[button-number{x}overlay-button-number]
[button-number...] • button-number—Number of a line button on an IP
phone.
Example: • separator—Single character that denotes the type of
Router(config-ephone)# button 1:10 2:11 3b12 characteristics to be associated with the button.
4o13,14,15
• dn-tag—Unique sequence number of the ephone-dn
that you want to appear on this button. For overlay lines
(separator is o or c), this argument can contain up to
25 ephone-dn tags, separated by commas.
• x—Separator that creates an overlay rollover button.
• overlay-button-number—Number of the overlay button
that should overflow to this button.
Example:
Router(config)# telephony-service
Step 9 create cnf-files Builds the XML configuration files that are required for
Cisco Unified CME phones.
Example:
Router(config-telephony)# create cnf-files
Step 10 restart {all [time-interval] | mac-address} Performs a fast reset of the specified phone or all phones
associated with this Cisco Unified CME router. Does not
contact the DHCP server.
Example:
Router(config-telephony)# restart all • all—Restarts all phones associated with a
Cisco Unified CME router.
• time-interval—(Optional) Time interval, in seconds,
between the beginning of each phone restart.
Range: 0 to 60. Default is 15.
• mac-address—Restarts the phone that has the specified
MAC address.
Step 11 end Exits to privileged EXEC mode.
Example:
Router(config-telephony)# end
Enabling a SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists
A watcher can monitor the status of lines associated with internal and external directory numbers
(presentities) through the BLF speed-dial and BLF call-list presence features. To enable the BLF
notification features on a SIP phone, perform the following steps.
Prerequisites
• Presence must be enabled on the Cisco Unified CME router. See the “Enabling Presence for Internal
Lines” section on page 800.
• A directory number must be enabled as a presentity with the allow watch command to provide BLF
status notification. See the “Enabling a Directory Number to be Watched” section on page 801.
• SIP phones must be configured with a directory number under voice register pool configuration
mode (use dn keyword in number command); direct line numbers are not supported.
Restrictions
BLF Call-List
• Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
BLF Speed-Dial
• Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. number tag dn dn-tag
5. blf-speed-dial tag number label string
6. presence call-list
7. exit
8. voice register global
9. mode cme
10. create profile
11. restart
12. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 1 to be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 number tag dn dn-tag Assigns a directory number to the SIP phone.
• tag—Identifier when there are multiple number
Example: commands. Range: 1 to 10.
Router(config-register-pool)# number 1 dn 2
• dn-tag—Directory number tag that was defined using
the voice register dn command.
Step 5 blf-speed-dial tag number label string Enables BLF monitoring of a directory number associated
with a speed-dial number on the phone.
Example: • tag—Number that identifies the speed-dial index.
Router(config-register-pool)# blf-speed-dial 3 Range: 1 to 7.
3001 label sales
• number—Telephone number to speed dial.
• string—Alphanumeric label that identifies the
speed-dial button. The string can contain a maximum of
30 characters.
Step 6 presence call-list Enables BLF monitoring of directory numbers that appear
in call lists and directories on this phone.
Example: • For a directory number to be monitored, it must have
Router(config-register-pool)# presence the allow watch command enabled.
call-list
• To enable BLF monitoring for call lists on all phones in
this Cisco Unified CME system, use this command in
presence mode. See the “Enabling Presence for Internal
Lines” section on page 800.
Step 7 exit Exits voice register pool configuration mode and enters
privileged EXEC mode.
Example:
Router(config-register-pool)# exit
Example:
Router(config-register-global)# end
Prerequisites
Presence service must be enabled for internal lines. See the “Enabling Presence for Internal Lines”
section on page 800.
SUMMARY STEPS
1. enable
2. configure terminal
3. presence
4. server ip-address
5. allow subscribe
6. watcher all
7. sccp blf-speed-dial retry-interval seconds limit number
8. exit
9. voice register global
10. authenticate presence
DETAILED STEPS
Example:
Router# configure terminal
Step 3 presence Enables presence service and enters presence configuration
mode.
Example:
Router(config)# presence
Step 4 server ip-address Specifies the IP address of a presence server for sending
presence requests from internal watchers to external
presentities.
Example:
Router(config-presence)# server 10.10.10.1
Step 5 allow subscribe Allows internal watchers to monitor external directory
numbers.
Example:
Router(config-presence)# allow subscribe
Step 6 watcher all Allows external watchers to monitor internal directory
numbers.
Example:
Router(config-presence)# watcher all
Step 7 sccp blf-speed-dial retry-interval seconds (Optional) Sets the retry timeout for BLF monitoring of
limit number speed-dial numbers on phones running SCCP.
• seconds—Retry timeout in seconds. Range: 60 to 3600.
Example: Default: 60.
Router(config-presence)# sccp blf-speed-dial
retry-interval 90 limit number 15 • number—Maximum number of retries.
Range: 10 to 100. Default: 10.
Step 8 exit Exits presence configuration mode.
Example:
Router(config-presence)# exit
Step 9 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Router(config)# voice register global
Example:
Router(config-register-global)# end
Step 3 show presence subscription [details | presentity telephone-number | subid subscription-id summary]
Use this command to display information about active presence subscriptions.
Router# show presence subscription summary
Troubleshooting Presence
Step 1 debug presence {all | asnl | errors | event | info | timer | trace | xml}
This command displays debugging information about the presence service.
Router# debug presence errors
Building configuration...
!
voice register dn 1
number 2101
allow watch
!
voice register dn 2
number 2102
allow watch
!
voice register pool 1
id mac 0015.6247.EF90
type 7971
number 1 dn 1
blf-speed-dial 1 1001 label "1001"
!
voice register pool 2
id mac 0012.0007.8D82
type 7912
number 1 dn 2
!
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 11.1.1.2 255.255.255.0
duplex full
speed 100
media-type rj45
no negotiation auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
media-type rj45
negotiation auto
!
ip route 0.0.0.0 0.0.0.0 11.1.1.1
!
ip http server
!
!
!
tftp-server flash:Jar41sccp.8-0-0-103dev.sbn
tftp-server flash:cvm41sccp.8-0-0-102dev.sbn
tftp-server flash:SCCP41.8-0-1-0DEV.loads
tftp-server flash:P00303010102.bin
tftp-server flash:P00308000100.bin
tftp-server flash:P00308000100.loads
tftp-server flash:P00308000100.sb2
tftp-server flash:P00308000100.sbn
tftp-server flash:SIP41.8-0-1-0DEV.loads
tftp-server flash:apps41.1-1-0-82dev.sbn
tftp-server flash:cnu41.3-0-1-82dev.sbn
tftp-server flash:cvm41sip.8-0-0-103dev.sbn
tftp-server flash:dsp41.1-1-0-82dev.sbn
tftp-server flash:jar41sip.8-0-0-103dev.sbn
tftp-server flash:P003-08-1-00.bin
tftp-server flash:P003-08-1-00.sbn
tftp-server flash:P0S3-08-1-00.loads
tftp-server flash:P0S3-08-1-00.sb2
tftp-server flash:CP7912080000SIP060111A.sbin
tftp-server flash:CP7912080001SCCP051117A.sbin
tftp-server flash:SCCP70.8-0-1-11S.loads
tftp-server flash:cvm70sccp.8-0-1-13.sbn
tftp-server flash:jar70sccp.8-0-1-13.sbn
tftp-server flash:SIP70.8-0-1-11S.loads
tftp-server flash:apps70.1-1-1-11.sbn
tftp-server flash:cnu70.3-1-1-11.sbn
tftp-server flash:cvm70sip.8-0-1-13.sbn
tftp-server flash:dsp70.1-1-1-11.sbn
tftp-server flash:jar70sip.8-0-1-13.sbn
!
control-plane
!
dial-peer voice 2001 voip
preference 2
destination-pattern 1...
session protocol sipv2
session target ipv4:11.1.1.4
dtmf-relay sip-notify
!
presence
server 11.1.1.4
sccp blf-speed-dial retry-interval 70 limit 20
presence call-list
max-subscription 128
watcher all
allow subscribe
!
sip-ua
authentication username jack password 021201481F
presence enable
!
!
telephony-service
load 7960-7940 P00308000100
load 7941GE SCCP41.8-0-1-0DEV
load 7941 SCCP41.8-0-1-0DEV
load 7961GE SCCP41.8-0-1-0DEV
load 7961 SCCP41.8-0-1-0DEV
load 7971 SCCP70.8-0-1-11S
load 7970 SCCP70.8-0-1-11S
load 7912 CP7912080000SIP060111A.sbin
max-ephones 100
max-dn 300
ip source-address 11.1.1.2 port 2000
url directories http://11.1.1.2/localdirectory
max-conferences 6 gain -6
call-forward pattern .T
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1 dual-line
number 2001
allow watch
!
!
ephone-dn 2 dual-line
number 2009
allow watch
application default
!
!
ephone-dn 3
number 2005
allow watch
!
!
ephone-dn 4 dual-line
number 2002
!
!
ephone 1
mac-address 0012.7F57.62A5
fastdial 1 1002
blf-speed-dial 1 2101 label "2101"
blf-speed-dial 2 1003 label "1003"
blf-speed-dial 3 2002 label "2002"
type 7960
button 1:1 2:2
!
!
!
ephone 3
mac-address 0015.6247.EF91
blf-speed-dial 2 1003 label "1003"
type 7971
button 1:3 2:4
!
!
!
line con 0
exec-timeout 0 0
password lab
stopbits 1
line aux 0
stopbits 1
line vty 0 4
password lab
login
!
scheduler allocate 20000 1000
!
end
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 46 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes ring tones features in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Ring Tones, page 819
• How to Configure Ring Tones, page 821
• Configuration Examples for Ring Tones, page 826
• Additional References, page 827
• Feature Information for Ring Tones, page 828
Distinctive Ringing
Distinctive ring is used to identify internal and external incoming calls. An internal calls is defined as a
call originating from any Cisco Unified IP phone that is registered in Cisco Unified CME or is routed
through the local FXS port.
In Cisco CME 3.4 and earlier versions, the standard ring pattern is generated for all calls to local SCCP
endpoints. In Cisco Unified CME 4.0, the following distinctive ring features are supported for SCCP
endpoints:
• Specify one of three ring patterns to be used for all types of incoming calls to a particular directory
number, on all phones on which the directory number appears. If a phone is already in use, an
incoming call is presented as a call-waiting call and uses a distinctive call-waiting beep.
• Specify whether the distinctive ring is used only if the incoming called number matches the primary
or secondary number defined for the ephone-dn. If no secondary number is defined for the
ephone-dn, the secondary ring option has no effect.
• Associate a feature ring pattern with a specific button on a phone so that different phones that share
the same directory number can use a different ring style.
For local SIP endpoints, the type of ring sound requested is signaled to the phone using an alert-info
signal. If distinctive ringing is enabled, Cisco Unified CME generates the alert-info for incoming calls
from any phone that is not registered in Cisco Unified CME, to the local endpoint. Alert-info from an
incoming leg can be relayed to an outgoing leg with the internally generated alert-info taking
precedence.
Cisco Unified IP phones use the standard Telcordia Technologies distinctive ring types.
On-Hold Indicator
On-hold indicator is an optional feature that generates a ring burst on idle IP phones that have placed a
call on hold. An option is available to generate call-waiting beeps for occupied phones that have placed
calls on hold. This feature is disabled by default. For configuration information, see the “SCCP:
Enabling On-Hold Indicator” section on page 824.
LED color display for hold state, also known as I-Hold, is supported in Cisco Unified CME 4.0(2) and
later versions. The I-Hold feature provides a visual indicator for distinguishing a local hold from a
remote hold on shared lines on supported phones, such as the Cisco Unified IP Phone 7931G. This
feature requires no additional configuration.
Prerequisites
Cisco Unified CME 4.0 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. ring {external | internal | feature} [primary | secondary]
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example:
Router(config)# ephone-dn 29
Example:
Router(config-ephone-dn)# number 2333
Step 5 ring {external | internal | feature} [primary | Designates which ring pattern to be used for all types of
secondary] incoming calls to this directory number, on all phones on
which the directory number appears.
Example:
Router(config-ephone-dn)# ring internal
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
Prerequisites
Cisco Unified CME 4.0 or a later version.
SUMMARY STEPS
DETAILED STEPS
Step 1 Create a PCM file for each customized ring tone (one ring per file). The PCM files must comply with
the following format guidelines.
• Raw PCM (no header)
• 8000 samples per second
• 8 bits per sample
• µLaw compression
• Maximum ring size—16080 samples
• Minimum ring size—240 samples
Step 3 Copy the PCM and XML files to system Flash on the Cisco Unified CME router. For example:
copy tftp://192.168.1.1/RingList.xml flash:
copy tftp://192.168.1.1/DistinctiveRingList.xml flash:
copy tftp://192.168.1.1/Piano1.raw flash:
copy tftp://192.168.1.1/Chime.raw flash:
Step 4 Use the tftp-server command to enable access to the files. For example:
tftp-server flash:RingList.xml
tftp-server flash:DistinctiveRingList.xml
tftp-server flash:Piano1.raw
tftp-server flash:Chime.raw
Step 5 Reboot the IP phones. After reboot, the IP phones download the XML and ring tone files. Select the
customized ring by pressing the Settings button followed by the Ring Type menu option on a phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. hold-alert timeout {idle | originator | shared}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example:
Router(config)# ephone-dn 20
Step 4 hold-alert timeout {idle | originator | Sets audible alert notification on the Cisco Unified IP phone for
shared} alerting the user about on-hold calls.
Note From the perspective of the originator of the call on hold,
Example: the originator and shared keywords provide the same
Router(config-ephone-dn)# hold-alert 15 functionality.
idle
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
Prerequisites
Cisco Unified CME 3.4 or a later version.
Restrictions
bellcore-dr1 to bellcore-dr5 are the only Telcordia options that are supported for SIP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. external-ring {bellcore-dr1 | bellcore-dr2 | bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 47 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the soft-key features in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Soft Keys, page 829
• How to Customize Soft Keys, page 832
• Configuration Examples for Soft-Keys, page 842
• Where to Go Next, page 844
• Additional References, page 844
• Feature Information for Soft Keys, page 845
already has a template applied to it, the second template overwrites the first phone template information.
The new information takes effect only after you generate a new configuration file and restart the phone,
otherwise the previously configured template remains in effect.
In Cisco Unified CME 4.1, customizing the soft key display for IP phones running SIP is supported only
for the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
For configuration information, see the “How to Customize Soft Keys” section on page 832.
Note If Cisco Unified CME does not receive a #, each account code digit is processed only after a timer
expires. The timer is 30 seconds for the first digit entered, then x seconds for each subsequent digit,
where x equals the number of seconds configured with the timeouts interdigit (telephony-service)
command. The default value for the interdigit timeout is 10 seconds. The account code digits do not
appear in show output until after being processed.
Feature Blocking
In Cisco Unified CME 4.0 and later versions, individual soft-key features can be blocked on one or more
phones. You specify the features that you want blocked by adding the features blocked command to an
ephone template. The template is then applied under ephone configuration mode to one or more ephones.
If a feature is blocked using the features blocked command, the soft key is not removed, but it does not
function. For configuration information, see the “Configuring Feature Blocking” section on page 840.
To remove a soft-key display, use the appropriate no softkeys command. See the “SCCP: Modifying
Soft-Key Display” section on page 832.
Prerequisites
• Cisco CME 3.2 or a later version.
• Cisco Unified 4.2 or a later version to enable soft keys during the ringing call state.
• The HLog soft key must be enabled with the hunt-group logout HLog command before it will be
displayed. For more information, see the “SCCP: Configuring Hunt Groups” section on page 596.
• The Flash soft key must be enabled with the fxo hook-flash command before it will be displayed.
For configuration information, see the “Enabling Flash Soft Key” section on page 838.
Restrictions
• The third soft-key button on the Cisco Unified IP Phone 7905G and Cisco Unified IP Phone 7912G
is reserved for the Message soft key. For these phones’ templates, the third soft-key defaults to the
Message soft key. For example, the softkeys idle Redial Dnd Pickup Login Gpickup command
configuration displays, in order, the Redial, DND, Message, PickUp, Login, and GPickUp soft keys.
• The NewCall soft key cannot be disabled on the Cisco Unified IP Phone 7905G or Cisco Unified IP
Phone 7912G.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-template template-tag
4. softkeys alerting {[Acct] [Callback] [Endcall]}
5. softkeys connected {[Acct] [Confrn] [Endcall] [Flash] [Hlog] [Hold] [Park] [Trnsfer]}
6. softkeys hold {[Newcall] [Resume]}
7. softkeys idle {[Cfwdall] [Dnd] [Gpickup] [Hlog] [Login] [Newcall] [Pickup] [Redial]}
8. softkeys seized {[Cfwdall] [Endcall] [Gpickup] [Hlog] [Pickup] [Redial]}
9. softkeys ringing {[Answer] [Dnd] [HLog]}
10. exit
11. ephone phone-tag
12. ephone-template template-tag
13. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an ephone
template.
Example: • template-tag—Unique identifier for the ephone template that
Router(config)# ephone-template 15 is being created. Range is 1 to 20.
Step 4 softkeys alerting {[Acct] [Callback] (Optional) Configures an ephone template for soft-key display
[Endcall]} during the alerting call state.
• You can enter any of the keywords in any order.
Example:
Router(config-ephone-template)# softkeys
• Default is all soft keys are displayed in alphabetical order.
alerting Callback Endcall • Any soft key that is not explicitly defined is disabled.
Step 5 softkeys connected {[Acct] [Confrn] (Optional) Configures an ephone template for soft-key display
[Endcall] [Flash] [Hlog] [Hold] [Park] during the call-connected state.
[Trnsfer]}
• You can enter any of the keywords in any order.
Example:
Router(config-ephone-template)# exit
Step 11 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 36
Step 12 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 13 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SCCP: Generating Configuration Files for SCCP Phones” section
on page 247.
Prerequisites
Cisco Unified CME 4.1 or a later version.
Restrictions
• This feature is supported only for Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G,
7961GE, 7970G, and 7971GE.
• You can download a custom soft key XML file from a TFTP server, however if the soft key XML
file contains an error, the soft keys might not work properly on the phone. We recommend the
following procedure for creating a soft key template in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register template template-tag
4. softkeys connected {[Confrn] [Endcall] [Hold] [Trnsfer]}
5. softkeys hold {[Newcall] [Resume]}
6. softkeys idle {[Cfwdall] [Newcall] [Redial]}
7. softkeys seized {[Cfwdall] [Endcall] [Redial]}
8. exit
9. voice register pool pool-tag
10. template template-tag
11. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to create
a SIP phone template.
Example: • template-tag—Range: 1 to 10.
Router(config)# voice register template 9
Step 4 softkeys connected {[Confrn] [Endcall] [Hold] (Optional) Configures an SIP phone template for soft-key
[Trnsfer]} display during the call-connected state.
• You can enter the keywords in any order.
Example:
Router(config-register-template)# softkeys
• Default is all soft keys are displayed in alphabetical
connected Endcall Hold Transfer order.
• Any soft key that is not explicitly defined is disabled.
Step 5 softkeys hold {[Newcall] {Resume]} (Optional) Configures a phone template for soft-key display
during the call-hold state.
Example: • Default is that the NewCall and Resume soft keys are
Router(config-register-template)# softkeys hold displayed in alphabetical order.
Resume
• Any soft key that is not explicitly defined is disabled.
Step 6 softkeys idle {[Cfwdall] [Newcall] [Redial]} (Optional) Configures a phone template for soft-key display
during the idle state.
Example: • You can enter the keywords in any order.
Router(config-register-template)# softkeys idle
Newcall Redial Cfwdall
• Default is all soft keys are displayed in alphabetical
order.
• Any soft key that is not explicitly defined is disabled.
Step 7 softkeys seized {[Cfwdall] [Endcall] [Redial]} (Optional) Configures a phone template for soft-key display
during the seized state.
Example: • You can enter the keywords in any order.
Router(config-register-template)# softkeys
seized Endcall Redial Cfwdall
• Default is all soft keys are displayed in alphabetical
order.
• Any soft key that is not explicitly defined is disabled.
Step 8 exit Exits voice register template configuration mode.
Example:
Router(config-register-template)# exit
Example:
Router(config-register-pool)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” section on
page 250.
number 1 dn 4
template 7
dialplan 3
!
Step 2 show telephony-service ephone-template
or
show voice register template template-tag
This command displays the contents of individual templates.
Router# show telephony-service ephone-template
ephone-template 1
softkey ringing Answer Dnd
conference drop-mode never
conference add-mode all
conference admin: No
Always send media packets to this router: No
Preferred codec: g711ulaw
User Locale: US
Network Locale: US
or
Router# show voice register template 7
Temp Tag 7
Config:
Attended Transfer is enabled
Blind Transfer is enabled
Semi-attended Transfer is enabled
Conference is enabled
Caller-ID block is disabled
DnD control is enabled
Anonymous call block is disabled
Voicemail is 52001, timeout 30
KPML is disabled
Transport type is tcp
softkey connected Endcall Trnsfer Confrn Hold
softkey hold Resume Newcall
softkey idle Newcall Redial Cfwdall
Restrictions
The IP phone must support soft-key display.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. fxo hook-flash
5. restart all
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 fxo hook-flash Enables the Flash soft key on phones that support soft-key
display, on PSTN calls using an FXO port.
Example: Note The Flash soft key display is automatically disabled
Router(config-telephony)# fxo hook-flash for local IP-phone-to-IP-phone calls.
Step 5 restart all Performs a fast reboot of all phones associated with this
Cisco Unified CME router. Does not contact the DHCP or
TFTP server for updated information.
Example:
Router(config-telephony)# restart all
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
telephony-service
fxo hook-flash
load 7960-7940 P00305000600
load 7914 S00103020002
max-ephones 100
max-dn 500
.
.
.
Step 2 Use the show telephony-service command to show only the telephony-service portion of the
configuration.
Prerequisites
Cisco Unified CME 4.0 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-template template-tag
4. features blocked [CFwdAll] [Confrn] [GpickUp] [Park] [PickUp] [Trnsfer]
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. restart
9. Repeat Step 5 to Step 8 for each phone to which the template should be applied.
10. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique sequence number that identifies
Example: this template during configuration tasks. Range is
Router(config)# ephone-template 1 1 to 20.
Example:
Router(config-ephone-template)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks. The maximum
Router(config)# ephone 25 number of ephones for a particular Cisco Unified CME
system is version- and platform-specific. For the range
of values, see the CLI help.
Step 7 ephone-template template-tag Applies an ephone template to an ephone.
• template-tag—Template number that you want to apply
Example: to this ephone.
Router(config-ephone)# ephone-template 1
Note To view your ephone-template configurations, use
the show telephony-service ephone-template
command.
Step 8 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example: Note If you are applying the template to more than one
Router(config-ephone)# restart ephone, you can use the restart all command in
telephony-service configuration mode to reboot all
the phones so they have the new template
information.
Step 9 Repeat Step 5 to Step 8 for each phone to which the —
template should be applied.
Step 10 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
ephone-template 2
softkeys idle Redial Newcall
softkeys seized Redial Endcall Pickup
softkeys alerting Redial Endcall
softkeys connected Endcall Hold Trnsfer
ephone 10
ephone-template 2
ephone 13
ephone-template 1
ephone 15
ephone-template 1
ephone 34
ephone-template 2
Modifying the HLog Soft Key for Ephone Hunt Groups: Example
The following example establishes the appearance and order of soft keys for phones that are configured
with ephone-template 7. The Hlog key is available when a phone is idle, when it has seized a line, or
when it is connected to a call. Phones without soft keys can use the standard HLog codes to toggle ready
and not-ready status.
telephony-service
hunt-group logout HLog
fac standard
.
.
ephone-template 7
softkeys connected Endcall Hold Transfer Hlog
softkeys idle Newcall Redial Pickup Cfwdall Hlog
softkeys seized Endcall Redial Pickup Cfwdall Hlog
ephone-dn 2
number 2333
ephone 3
button 1:2
ephone-template 1
ephone-dn 78
number 2579
ephone 3
ephone-template 1
mac-address C910.8E47.1282
type anl
button 1:78
Where to Go Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. For more information, see “Generating Configuration Files for Phones” on
page 245.
Ephone Templates
The softkeys commands are included in ephone templates that are applied to one or more individual
ephones. For more information about templates, see “Creating Templates” on page 881.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 48 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the speed dial support available in Cisco Unified Communications Manager
Express (Cisco Unified CME).
Contents
• Information About Speed Dial, page 847
• How to Configure Speed Dial, page 851
• Configuration Examples for Speed Dial, page 863
• Where to Go Next, page 864
• Additional References, page 865
• Feature Information for Speed Dial, page 866
Bulk speed-dial lists contain entries of speed-dial codes and the associated phone numbers to dial. Each
entry in a speed-dial list must appear on a separate line. The fields in each entry are separated by
commas (,). A line that begins with a semicolon (;) is handled as a comment. The format of each entry
is shown in the following line.
index,digits,[name],[hide],[append]
Table 50 explains the fields in a bulk speed-dial list entry.
Field Description
index Zero-filled number that uniquely identifies this index entry.
Maximum length: 4 digits. All index entries must be the same
length.
digits Telephone number to dialed. Represents a fully qualified
E.164 number. Use a comma (,) to represent a one-second
pause.
name (Optional) Alphanumeric string to identify a name, up to 30
characters.
hide (Optional) Enter hide to block the display of the dialed
number.
append (Optional) Enter append to allow additional digits to be
appended to this number when dialed.
To place a call to a speed-dial entry in a list, the phone user must first dial a prefix, followed by the list
ID number, then the index for the bulk speed-dial list entry to be called.
For configuration information, see the “SCCP: Enabling Bulk-Loading Speed-Dial” section on
page 858.
The following example shows a monitor-line configuration. Extension 2311 is the manager’s line, and
ephone 1 is the manager’s phone. The manager’s assistant monitors extension 2311 on button 2 of
ephone 2. When the manager is on the line, the lamp is lit on the assistant’s phone. If the lamp is not lit,
the assistant can speed-dial the manager by pressing button 2.
ephone-dn 11
number 2311
ephone-dn 22
number 2322
ephone 1
button 1:11
ephone 2
button 1:22 2m11
No additional configuration is required to enable a phone user to speed dial the number of a monitored
shared line, when the monitored line is in an idle call state.
Prerequisites
An XML file called speeddial.xml must be created and copied to the TFTP server application on the
Cisco Unified CME router. The contents of speeddial.xml must be valid as defined in the Cisco specified
directory DTD. See the Cisco IP Phone Services Application Development Notes.
Restrictions
• If a speed dial XML file contains incomplete information, for example the name or telephone
number is missing for an entry, any information in the file that is listed after the incomplete entry is
not displayed when the local speed dial directory option is used on a phone.
• Before Cisco CME 4.1, local speed-dial menu is not supported on SIP phones.
• Before Cisco CME 3.3, analog phones are limited to nine speed-dial numbers.
SUMMARY STEPS
1. enable
2. copy tftp flash
3. configure terminal
4. ip http server
5. ip http path flash:
6. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 4 ip http server Enables the Cisco web-browser user interface on the router.
Example:
Router(config)# ip http server
Step 5 ip http path flash: Sets the base HTTP path to flash memory.
Example:
Router(config)# ip http path flash:
Step 6 exit Returns to privileged EXEC mode.
Example:
Router(config)# exit
Prerequisites
Cisco Unified CME 4.0(2) or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service dss
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 service dss Configures DSS (Direct Station Select) service globally
for all phone users in Cisco Unified CME.
Example:
Router(config-telephony)# service dss
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end
Restrictions
• A personal speed-dial menu is available only on Cisco Unified IP Phones 7940, 7960, 7960G,
7970G and 7971G-GE.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. fastdial dial-tag number name name-string
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number for the phone for
Example: which you want to program personal speed-dial
Router(config)# ephone 1 numbers.
Example:
Router(config-ephone)# end
Restrictions
• On-hook abbreviated dialing using the Abbr soft key is supported only on the following phone
types:
– Cisco Unified IP Phone 7905G
– Cisco Unified IP Phone 7912G
– Cisco Unified IP Phone 7920G
– Cisco Unified IP Phone 7970G
– Cisco Unified IP Phone 7971G-GE
• System-level speed-dial codes cannot be changed by the phone user, at the phone.
• Before Cisco CME 3.3, analog phones were limited to nine speed-dial numbers.
• Before to Cisco CME 3.3, speed-dial entries that were in excess of the number of physical phone
buttons available were ignored by IP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. speed-dial speed-tag digit-string [label label-text]
5. exit
6. telephony-service
7. directory entry {directory-tag number name name | clear}
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone on which you are adding speed-dial
Router(config)# ephone 55 capability.
Step 4 speed-dial speed-tag digit-string [label Defines a unique speed-dial identifier, a digit string to dial,
label-text] and an optional label to display next to the button.
• speed-tag—Identifier for a speed-dial definition.
Example: Range is 1 to 33.
Router(config-ephone)# speed-dial 1 +5001 label
“Head Office”
Step 5 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-ephone)# restart
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-ephone)# exit
Step 7 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 8 directory entry {{directory-tag number name Adds a system-level directory and speed-dial definition.
name} | clear}
• directory-tag—Digit string that provides a unique
identifier for this entry. Range is 1 to 99.
Example:
Router(config-telephony)# directory entry 45
• If the same tags 1 through 33 are configured at a
8185550143 name Corp Acctg phone-level by using speed-dial command, and at a
system-level by using this command, the local
definition takes precedence. To prevent this conflict,
we recommend that you use only codes 34 to 99 for
system-level speed-dial numbers.
Example:
Router(config-telephony)# end
Prerequisites
• Cisco Unified CME 4.0 or a letter version.
• The bulk speed-dial text files containing the lists must be available in a location that is available to
the Cisco Unified CME router: flash, slot, or TFTP location.
Restrictions
• Bulk speed dial is not supported on FXO trunk lines.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. bulk-speed-dial list list-id location
5. bulk-speed-dial prefix prefix-code
6. exit
7. ephone phone-tag
8. bulk-speed-dial list list-id location
9. end
DETAILED STEPS
Example:
Router# configure terminal
Example:
Router(config)# telephony-service
Step 4 bulk-speed-dial list list-id location Identifies the location of a bulk speed-dial list.
• list-id—Digit that identifies the list to be used. Range is
Example: 0 to 9.
Router(config-telephony)# bulk-speed-dial list 6
flash:sd_dept_0_1_8.txt
• location—Location of the bulk speed-dial text file in
URL format. Valid storage locations are TFTP, Slot 0/1,
and flash memory.
Step 5 bulk-speed-dial prefix prefix-code Sets the prefix code that phone users dial to access speed-dial
numbers from a bulk speed-dial list.
Example: • prefix-code—One- or two-character access code for
Router(config-telephony)# bulk-speed-dial prefix speed dial. Valid characters are digits from 0 to 9,
#7 asterisk (*), and pound sign (#). Default is #.
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-telephony)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 25
Step 8 bulk-speed-dial list list-id location Identifies the location of a bulk speed-dial list.
• list-id—Digit that identifies the list to be used. Range is
Example: 0 to 9.
Router(config-ephone)# bulk-speed-dial list 7
flash:lmi_sd_list_08_24_95.txt
• location—Location of the bulk speed-dial text file in
URL format. Valid storage locations are TFTP, Slot 0/1,
and flash memory.
Step 9 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
Prerequisites
Cisco CME 3.4 or a later version.
Restrictions
• Certain SIP IP phones, such as the Cisco Unified IP Phone 7960 and 7940, cannot be configured to
enable speed dialing. Phone users with these phones must manually configure speed-dial numbers
by using the user interface at their Cisco Unified IP phone.
• On Cisco Unified IP phones, speed-dial definitions are assigned to available buttons that have not
been assigned to actual extensions. Speed-dial definitions are assigned in the order of their identifier
numbers.
• Phones with Cisco ATA devices are limited to a maximum of nine speed-dial numbers. Speed-dial
numbers cannot be programmed by using the user interface at the phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. speed-dial speed-tag digit-string [label label-text]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
parameters for specified SIP phone.
Example:
Router(config)# voice register pool 23
Step 4 speed-dial speed-tag digit-string [label Creates a speed-dial definition in Cisco Unified CME for a
label-text] SIP phone or analog phone that uses an analog adapter
(ATA).
Example: • speed-tag—Unique sequence number that identifies the
router(config-register-pool)# speed-dial 2 speed-dial definition during configuration. Range is 1
+5001 label “Head Office”
to 5.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end
Examples
The following example shows how to set speed-dial button 2 to dial the head office at extension 5001
and locks the setting so that the phone user cannot change the setting at the phone:
Router(config)# voice register pool 23
Router(config-register-pool)# speed-dial 2 +5001 label “Head Office”
Prerequisites
• Cisco Unified CME 4.1 or a later version.
Restrictions
• Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE—Personal
speed-dial numbers can only be created in Cisco Unified CME, using this procedure.
• Cisco Unified IP Phone 7905, 7912, 7940, and 7960—Speed dial numbers can only be created by
the user directly on the phone and not in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. fastdial dial-tag number [name name-string]
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config-register-pool)# voice register to be configured. Range is version and
pool 1 platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 fastdial dial-tag number [name name-string] Creates a personal speed-dial number on this SIP phone.
• dial-tag—Unique number to identify this entry during
Example: configuration. Range: 1 to 24.
Router(config-register-pool)# fastdial 1 5552
name Sales
• number—Telephone number or extension to be dialed.
• name name-string—(Optional) Label to appear in the
Personal Speed Dial menu, containing a string of a
maximum of 24 alphanumeric characters. Personal
speed dial is handled through an XML request, so
characters that have special meaning to HTTP, such as
ampersand (&), percent sign (%), semicolon (;), angle
brackets (< >), and vertical bars (||), are not allowed.
• Repeat this command for each personal speed-dial
number that you want to create on this phone.
Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-register-pool)# end
The following XML file—speeddial.xml, defines three speed-dial numbers that will appear to the user
after they press the Directories button on an IP phone.
<CiscoIPPhoneDirectory>
<Title>Local Speed Dial</Title>
<Prompt>Record 1 to 1 of 1 </Prompt>
<DirectoryEntry>
<Name>Security</Name>
<Telephone>71111</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Marketing</Name>
<Telephone>71234</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Tech Support</Name>
<Telephone>71432</Telephone>
</DirectoryEntry>
</CiscoIPPhoneDirectory>
telephony-service
directory entry 34 5003 name Accounting
directory entry 45 8185550143 name Corp Acctg
ephone-dn 3
number 2555
ephone-dn 4
number 2557
ephone 25
button 1:3 2:4
bulk-speed-dial list 7 flash:lmi_sd_list_08_24_95.txt
Where to Go Next
If you are finished creating or modifying speed-dial configurations for individual phones, you must
reboot phones to download the modified configuration. See “Resetting and Restarting Phones” on
page 257.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 51 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the video support for SCCP-based endpoints in Cisco Unified Communications
Manager Express (Cisco Unified CME).
Contents
• Prerequisites for Video Support for SCCP-Based Endpoints, page 867
• Information About Video Support for SCCP-Based Endpoints, page 869
• How to Configure Video for SCCP-Based Endpoints, page 872
• Additional References, page 878
• Feature Information for Video Support for SCCP-Based Endpoints, page 880
From a PC with Cisco Unified Video Advantage version 1.02 or later installed, ensure that the line
between the Cisco Unified Video Advantage and the Cisco Unified IP phone is green. For more
information, see the Cisco Unified Video Advantage User Guide.
• Ensure that the correct video firmware is installed on the Cisco Unified IP phone. Use the show
ephone phone-load command to view current ephone firmware. The following lists the minimum
firmware version for video-enabled Cisco Unified IP phones:
– Cisco Unified IP Phone 7940G 6.0(4)
– Cisco Unified IP Phone 7960G 6.0(4)
– Cisco Unified IP Phone 7970G 7.0(3)
– Cisco Unified IP Phone 7941G 7.0(3)
– Cisco Unified IP Phone 7961G 7.0(3)
Note Other video-enabled endpoints, if registered with Cisco Unified Communications Manager,
can place a video call to one of the Cisco Unified IP phones listed above if it is registered
with Cisco Unified CME.
Note After video is enabled globally, all video-capable ephones display the video icon.
Note The endpoint-capability match is executed each time a new call is set up or an existing call is resumed.
Note During an audio-only connection, all video-related media messages are skipped.
• With flow-through mode, the video media path is the same as for an audio call. Media packets flow
through the gateway, thus hiding the networks from each other.
Use the show voip rtp connection command to display information about RTP named-event packets,
such as caller-ID number, IP address, and port for both the local and remote endpoints, as show in the
following sample output.
Router# show voip rtp connections
Note For more information on slow-connect procedures, see Configuring Quality of Service for Voice.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. h323
5. call start slow
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode.
Example:
Router(config)# voice service voip
Step 4 h323 Enters H.323 voice-service configuration mode.
Example:
Router(config-voi-serv)# h323
Step 5 call start slow Forces an H.323 gateway to use slow-connect procedures
for all VoIP calls.
Example:
Router(config-serv-h323)# call start slow
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-serv-h323)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service phone videoCapability [0 | 1]
5. end
DETAILED STEP
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 service phone videoCapability [0 | 1] Enables or disables video capabilities for
Cisco Unified CME.
Example: • 0—(Optional) Disables capabilities.
Router(config-telephony)# service phone
videoCapability 1
• 1—(Optional) Enables capabilities.
Note This command is case sensitive. If it is not entered
exactly as shown, Cisco Unified CME accepts the
command, but video capabilities are not enabled.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
Prerequisites
Use the show ephone registered command to display which phones are registered to
Cisco Unified CME and have video capability. The following example shows that ephone 1 has video
capabilities and ephone 2 is an audio-only phone.
Router# show ephone registered
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. video
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies an
Example: ephone during configuration tasks. The maximum
Router(config)# ephone 6 number is platform-dependent.
Step 4 video Enables video capabilities on the specified ephone. Repeat
as necessary for all ephones that require video enabled.
Example:
Router(config-ephone)# video
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. video
5. maximum bit-rate value
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 video Enters video configuration mode.
Example:
Router(config-telephony)# video
Example:
Router(conf-tele-video)# end
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note The following table lists the Cisco Unified CME version that introduced support for a given feature.
Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes templates support available in Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Information About Templates, page 881
• How to Configure Templates, page 882
• Configuration Examples for Creating Templates, page 888
• Where to Go Next, page 889
• Additional References, page 889
• Feature Information for Creating Templates, page 891
Phone Templates
An ephone or voice-register template is a set of features that can be applied to one or more individual
phones using a single command.
Ephone templates were introduced in Cisco CME 3.2 to manipulate soft-key display and order on IP
phones. In Cisco Unified CME 4.0, ephone templates were significantly enhanced to include a number
of additional phone features. Templates allow you to uniformly and easily implement the features you
select for a set of phones. A maximum of 20 ephone templates can be created in a Cisco Unified CME
system, although an ephone can have only one template applied to it at a time.
If you use an ephone template to apply a command to a phone and you also use the same command in
ephone configuration mode for the same phone, the value set in ephone configuration mode has priority.
Voice-register templates were introduced in Cisco CME 3.4 to enable sets of features to be applied to
individual SIP IP phones that are connected directly in Cisco Unified CME. Typically, features to be
enabled by using a voice-register template are not configurable in other configuration modes. A
maximum 10 voice-register templates can be defined in Cisco Unified CME, although a phone can have
only one template applied to it at a time.
Type ? in ephone-template or voice-register-template configuration mode to display a list of features that
can be implemented by using templates.
Ephone-dn Templates
Ephone-dn templates allow you to apply a standard set of features to ephone-dns. A maximum of 15
ephone-dn templates can be created in a Cisco Unified CME system, although an ephone-dn can have
only one template applied to it at a time.
If you use an ephone-dn template to apply a command to an ephone-dn and you also use the same
command in ephone-dn configuration mode for the same ephone-dn, the value that you set in ephone-dn
configuration mode has priority.
Note The features that can be implemented using ephone-dn templates are available in the CLI help by
entering a question mark:
Router(config)# ephone-dn-template 1
Router(config-ephone-dn-template)# ?
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-template template-tag
4. command
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. restart
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an
ephone template.
Example: • template-tag—Unique identifier for the ephone
Router(config)# ephone-template 15 template that is being created. Range is 1 to 20.
Step 4 command Applies the specified command to the ephone template that
is being created. See the CLI help for a list of commands
that can be used in this step. Repeat this step for each
Example:
Router(config-ephone-template)# features
command that you want to add to the ephone template.
blocked Park Trnsfer
Step 5 exit Exits ephone-template configuration mode.
Example:
Router(config-ephone-template)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks.
Router(config)# ephone 36
Example:
Router(config-ephone)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn-template template-tag
4. command
5. exit
6. ephone-dn dn-tag
7. ephone-dn-template template-tag
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn-template template-tag Enters ephone-dn-template configuration mode to create an
ephone-dn template.
Example: • template-tag—Unique identifier for the ephone-dn
Router(config)# ephone-dn-template 3 template that is being created. Range is 1 to 20.
Example:
Router(config-ephone-dn-template)# exit
Step 6 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies this
Example: ephone-dn during configuration tasks.
Router(config)# ephone-dn 23
Step 7 ephone-dn-template template-tag Applies an ephone-dn template to the ephone-dn that is
being configured.
Example:
Router(config-ephone-dn)# ephone-dn-template 3
Step 8 end Returns to privileged EXEC mode.
Example:
Router(config-ephone-dn)# end
SUMMARY STEPS
DETAILED STEPS
Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register template template-tag
4. command
5. exit
6. voice register pool pool-tag
7. template template-tag
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to define
a template of common parameters for SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register template 1 • Range is 1 to 5.
Step 4 command Applies the specified command to this template and enables
the corresponding feature on any supported SIP phone that
uses a template in which this command is configure.
Example:
Router(config-register-template)# anonymous • Type ? to display list of commands that can be used in
block a voice register template.
• Repeat this step for each feature to be added to this
voice register template.
Step 5 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-template)# exit
Step 6 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example: • pool-tag—Unique sequence number of the Cisco SIP
Router(config)# voice register pool 3 phone to be configured. Range is 1 to 100 or the upper
limit as defined by max-pool command.
Step 7 template template-tag Applies a template created with the voice register template
command.
Example: • template-tag—Unique sequence number of the
Router(config-register-pool)# voice register template to be applied to the SIP phone specified by the
pool 1 voice register pool command. Range is 1 to 5.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end
Examples
The following example shows templates 1 and 2 and how to do the following:
• Apply template 1 to SIP phones 1 to 3
• Apply template 2 to SIP phone 4
• Remove a previously created template 5 from SIP phone 5.
Router(config)# voice register template 1
Router(config-register-temp)# anonymous block
Router(config-register-temp)# caller-id block
Router(config-register-temp)# voicemail 5001 timeout 15
Using Ephone Template to Block The Use of Park and Transfer Soft Keys
The following example creates an ephone template to block the use of Park and Transfer soft keys. It is
applied to ephone 36 and extension 2333.
ephone-template 15
features blocked Park Trnsfer
ephone-dn 2
number 2333
ephone 36
button 1:2
ephone-template 15
ephone-dn 23
number 2323
ephone-dn-template 3
ephone-dn 33
number 3333
ephone-dn-template 3
ephone 13
button 1:23
ephone 14
button 1:33
Where to Go Next
Soft-Key Display
The display of soft keys during different call states is managed using ephone templates. For more
information, see “Customizing Soft Keys” on page 829.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 53 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the screen and button features available for Cisco Unified IP phones connected
to Cisco Unified Communications Manager Express (Cisco Unified CME).
Contents
• Information About Cisco Unified IP Phone Options, page 893
• How to Configure Cisco Unified IP Phone Options, page 896
• Configuration Examples for Cisco Unified IP Phone Options, page 913
• Additional References, page 915
• Feature Information for Cisco Unified IP Phone Options, page 917
Content lines
Service window
Phone Labels
Pone labels are configurable text strings that can be displayed instead of extension numbers next to line
buttons on a Cisco Unified IP phone. By default, the number that is associated to a directory number,
and assigned to a phone, is displayed next to the applicable button. The label feature allows you to enter
a meaningful text string for each directory number so that a phone user with multiple lines can select a
line by label instead of by phone number, thus eliminating the need to consult in-house phone directories.
For configuration information, see the “SCCP: Creating Labels for Directory Numbers” section on
page 901 or the “SIP: Creating Labels for Directory Numbers” section on page 903.
For configuration information at the system level, see the “SCCP: Modifying Vendor Parameters for All
Phones” section on page 910. For configuration information for individual phones, see the “SCCP:
Modifying Vendor Parameters For a Specific Phone” section on page 911.
Prerequisites
Cisco Unified CME 4.0(2) or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone template template-tag
4. button-layout set phone-type [1 | 2]
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an
ephone template.
Example:
Router(config)# ephone-template 15
Example:
Router(config)# ephone 1
Step 7 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
Prerequisites
Directory number to be modified is already configured. For configuration information, see “SCCP:
Creating Directory Numbers” on page 158.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. description display-text
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
Example:
Router(config)# ephone-dn 55
Step 4 description display-text Defines a description for the header bar of a display-capable IP
phone on which this ephone-dn appears as the first line.
Example: • display-text—Alphanumeric character string, up to
Router(config-ephone-dn)# description 40 characters. String is truncated to 14 characters in the
408-555-0134 display.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
Prerequisites
• Cisco CME 3.4 or a a later version.
Restrictions
• This feature is supported only on Cisco Unified IP Phone 7940, 7940G, 7960, and 7960G.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. description string
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 description string Defines a customized description that appears in the header
bar of supported Cisco Unified IP phones
Example: • Truncated to 14 characters in the display.
Router(config-register-pool)# description
408-555-0100
• If string contains spaces, enclose the string in quotation
marks.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” on
page 250.
ephone-dn 1 dual-line
number 150 secondary 151
description 555-0150
call-forward busy 160
call-forward noan 160 timeout 10
huntstop channel
no huntstop
!
!
!
voice-register dn 1
number 1101
description 555-0101
ephone-dn 22
number 2149
description 408-555-0149
ephone 34
mac-address 0030.94C3.F96A
button 1:22 2:23 3:24
speed-dial 1 5004
speed-dial 2 5001
Prerequisites
Directory number for which the label is to be created is already configured. For configuration
information, see “SCCP: Creating Directory Numbers” on page 158.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. label label-string
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies the
Example: ephone-dn to which the label is to be associated.
Router(config)# ephone-dn 1
Step 4 label label-string Creates a custom label that is displayed on the phone next
to the line button that is associated with this ephone-dn. The
custom label replaces the default label, which is the number
Example:
Router(config-ephone-dn)# label user1
that was assigned to this ephone-dn.
• label-string—String of up to 30 alphanumeric
characters that provides the label text.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-ephone)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.
Prerequisites
• Cisco CME 3.4 or a later version.
• Directory number for which the label is to be created is already configured and must already have a
number assigned by using the number (voice register dn) command. For configuration
information, see “SIP: Creating Directory Numbers” on page 162.
Restrictions
• Only one label is permitted per directory number.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. label string
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config-register-global)# voice register
dn 17
Step 4 number number Defines a valid number for a directory number.
Example:
Router(config-register-dn)# number 7001
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” on
page 250.
Verifying Labels
Step 1 Use the show running-config command to verify your configuration. Descriptions for directory
numbers are listed in the ephone-dn and voice-register dn portions of the output.
Router# show running-config
ephone-dn 1 dual-line
number 150 secondary 151
label MyLine
call-forward busy 160
call-forward noan 160 timeout 10
huntstop channel
no huntstop
!
!
!
voice-register dn 1
number 1101
label MyLine
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. system message text-message
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)#
Step 4 system message text-message Defines a text message to display when a phone is idle.
• text-message—Alphanumeric string to display. Display uses
Example: proportional-width font, so the number of characters that are
Router(config-telephony)# system message displayed varies based on the width of the characters that are
ABC Company used. The maximum number of displayed characters is
approximately 30.
Step 5 url idle url idle-timeout seconds Defines the location of a file to display on phones that are not in
use and specifies the interval between refreshes of the display, in
seconds.
Example:
Router(config-telephony)# url idle • url—Any URL that conforms to RFC 2396.
http://www.abcwrecking.com/public/logo
idle-timeout 35 • seconds—Time interval between display refreshes, in
seconds. Range is 0 to 300.
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
What to Do Next
After configuring the url idle command, you must reset phones. See “SCCP: Using the reset Command”
on page 259.
telephony-service
fxo hook-flash
load 7960-7940 P00307020300
load 7914 S00104000100
max-ephones 100
max-dn 500
ip source-address 10.153.13.121 port 2000
max-redirect 20
timeouts ringing 100
system message XYZ Company
voicemail 7189
max-conferences 8 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
multicast moh 239.10.10.1 port 2000
web admin system name server1 password server1
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 92......
transfer-pattern 91..........
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern 99......
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
Restrictions
• Operation of these services is determined by the Cisco Unified IP phone capabilities and the content
of the specified URL.
• Provisioning a URL to access help screens using the i or ? buttons on a phone is not supported.
• Provisioning the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. url {directories | information | messages | services} url
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)#
Example:
Router(config-telephony)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.
Prerequisites
• Cisco CME 3.4 or a later version.
Restrictions
• Operation of these services is determined by the Cisco Unified IP phone capabilities and the content
of the specified URL.
• Provisioning a URL is supported only for Services and Directories feature buttons on SIP phones.
• Programmable Directories and Services feature buttons are supported only on the Cisco Unified IP
Phone 7960, 7960G, 7940, and 7940G.
• Provisioning the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register global Enters telephony-service configuration mode.
Example:
Router(config)#
Step 4 url {directory | service} url Associates a URL with the programmable feature buttons on SIP
phones.
Example:
Router(config-register-global)# url
directory http://10.0.0.11/localdirectory
Router(config-register-global)# url
service
http://10.0.0.4/CCMUser/123456/urltest.ht
ml
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-register-global)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” on
page 250.
Restrictions
• Only the parameters supported by the currently loaded firmware are available.
• The number and type of parameters may vary from one firmware version to the next.
• Only those parameters that are supported by a Cisco Unified IP phone and firmware version are
implemented. Parameters that are not supported are ignored.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service phone parameter-name parameter-value
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Example:
Router(config-telephony)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.
Restrictions
• Cisco Unified CME 4.0 or a later version.
• System must be configured to for per-phone configuration files. For configuration information, see
“SCCP: Defining Per-Phone Configuration Files and Alternate Location” on page 129.
• Only the parameters supported by the currently loaded firmware are available.
• The number and type of parameters may vary from one firmware version to the next.
• Only those parameters that are supported by a Cisco Unified IP phone and firmware version are
implemented. Parameters that are not supported are ignored.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone template template-tag
4. service phone parameter-name parameter-value
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an
ephone template.
Example:
Router (config)# ephone-template 15
Step 4 service phone parameter-name parameter-value Sets parameters for all IP phones that support the
configured functionality and to which this template is
applied.
Example:
Router(config-ephone-template)# service phone • The parameter name is word and case-sensitive. See
daysBacklightNotActive 1,2,3,4,5,6,7 the Cisco Unified CME Command Reference for a
Router(config-ephone-template)# service phone
list of parameters.
backlightOnTime 07:30
Router(config-ephone-template)# service phone • This command can also be configured in
backlightOnDuration 10:00 telephony-service configuration mode. For
Router(config-ephone-template)# service phone
backlightIdleTimeout 00.01
individual phones, the template configuration for
this command overrides the system-level
configuration for this command.
Step 5 exit Exits from this command mode to the next highest mode
in the configuration mode hierarchy.
Example:
Router(config-ephone-template)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.
Example:
Router(config)# ephone 1
Step 7 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end
What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.
tftp-server system:/its/SEPDEFAULT.cnf
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEP00036B54BB15.cnf.xml
tftp-server system:/its/germany/7960-font.xml alias German_Germany/7960-font.xml
tftp-server system:/its/germany/7960-dictionary.xml alias
German_Germany/7960-dictionary.xml
tftp-server system:/its/germany/7960-kate.xml alias German_Germany/7960-kate.xml
tftp-server system:/its/germany/SCCP-dictionary.xml alias
German_Germany/SCCP-dictionary.xml
tftp-server system:/its/germany/7960-tones.xml alias Germany/7960-tones.xml
Step 4 Use the debug tftp events command to verify that the phone is accessing the file when you reboot the
phone.
ephone-dn 2
number 2002
label Engineering
ephone-dn 56
number 2150
ephone 12
button 1:55 2:56
In the following example, the PC port is disabled on phones 26 and 27. All other phones have the PC
port enabled.
ephone-template 8
service phone pcPort 1
!
!
ephone 26
mac-address 1111.1111.1001
ephone-template 8
type 7960
button 1:26
!
!
ephone 27
mac-address 1111.2222.2002
ephone-template 8
type 7960
button 1:27
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 54 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
Contents
• Information About Interoperability with External Services, page 919
• How to Configure Interoperability with External Services, page 921
• Configuration Examples for Interoperability with Unified CCX, page 930
• Where to Go Next, page 939
• Additional References, page 940
• Feature Information for Interoperability with External Services, page 941
Table 55 Tasks to Configure Interoperability between Cisco CRS and Cisco Unified CME
Tip When setup launches, you are asked for the AXL
user ID and password that you created in
Cisco Unified CME. You also need to enter the
router IP address.
6 Configure CME Telephony Subsystem to enable Cisco CRS Administration Guide
interoperability with Unified CCX. at
7 Create users and assign the agent capability in Cisco http://www.cisco.com/en/US/prod
CRS. ucts/sw/custcosw/ps1846/prod_m
aintenance_guides_list.html.
Note A single Cisco Unified CME can support multiple session managers.
Prerequisites
• Cisco Unified CME 4.2 using Cisco IOS Release 12.4(11)XW2 or a later version.
• XML API must be configured to create a username for Unified CCX access. For configuration
information, see “Configuring the XML API” on page 959. Make note of the user ID, password, and
router’s IP address for using during the initial setup of Cisco CRS for Cisco Unified CME.
• Phones to be connected in Cisco Unified CME must be configured. When configuring a
Unified CCX agent phone, use the keep-conference endcall command to enable conference
initiators to exit from conference calls and end the conference for the remaining parties. For
configuration information, see “Configuring Conferencing” on page 647.
• The Cisco Unified CME router must be configured to accept incoming presence requests. For
configuration information, see “Configuring Presence Service” on page 797.
Restrictions
• Interoperability between Cisco Unified CME and Unified CCX is restricted to one Unified CCX per
Cisco Unified CME.
• Support for Multi-Party Ad Hoc and Meet-Me Conferencing features is not provided.
• Only incoming calls from PSTN trunk are supported for deployment of the interoperability feature.
Other trunks, such as SIP and H.323, are supported as usual in Cisco Unified CME, however, not for
customer calls to Unified CCX.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice call send-alert
4. voice service voip
5. callmonitor
6. gcid
7. allow-connections sip-to-sip
8. no supplementary-service sip moved-temporary
9. no supplementary-service sip refer
10. sip
11. registrar server [expires [max sec] [min sec]
12. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice call send-alert Enables the terminating gateway to send an alert message
instead of a progress message after it receives a call setup
message.
Example:
Router(config)# voice call send-alert
Step 4 voice service voip Enters voice-service configuration mode and specifies
voice-over-IP encapsulation.
Example:
Router(config)# voice service voip
Step 5 callmonitor Enables call monitoring messaging functionality.
• Used by Unified CCX for processing and reporting.
Example:
Router(config-voi-serv)# callmonitor
Step 6 gcid Enables Global Call-ID (Gcid) for call control purposes.
• Used by Unified CCX for tracking call.
Example:
Router(config-voi-serv)# gcid
Step 7 allow-connections sip-to-sip Allows connections between specific types of endpoints in
a VoIP network.
Example:
Router(config-voi-serv)# allow-connections
sip-to-sip
Step 8 no supplementary-service sip moved-temporary Prevents the router from sending a redirect response to the
destination for call forwarding.
Example:
Router(config-voi-serv)# no
supplementary-service sip moved-temporary
Step 9 no supplementary-service sip refer Prevents the router from forwarding a REFER message to
the destination for call transfers.
Example:
Router(config-voi-serv)# no
supplementary-service sip refer
Step 10 sip Enters SIP configuration mode.
Example:
Router(config-voi-srv)# sip
Prerequisites
• Up to eight session managers must be configured in Cisco Unified CME.
• Directory numbers associated with Unified CCX agent phones must be configured. Directory
numbers for agent phones must be configured as dual lines to allow an agent to make two call
connections at the same time using one phone line button. For configuration information, see
“Configuring Phones to Make Basic Calls” on page 147.
Restrictions
• Only SCCP phones can be configured as agent phones in Cisco Unified CME. The Cisco VG224
Analog Phone Gateway and analog and SIP phones are supported as usual in Cisco Unified CME,
however, not as Unified CCX agent phones.
• Cisco Unified IP Phone 7931 cannot be configured as an agent phone in Cisco Unified CME.
Cisco Unified IP Phone 7931s are supported as usual in Cisco Unified CME, however, not as
Unified CCX agent phones.
• Shared-line appearance is not supported on agent phones. A directory number cannot be associated
with more than one physical agent phone at one time.
• Overlaid lines are not supported on agent phones. More than one directory number cannot be
associated with a single line button on an agent phone.
• Monitored mode for a line button is not supported on agent phones. An agent phone cannot be
monitored by another phone.
• For call forward and call pickup, the directory number of an agent cannot forward to a Cisco CRS
route point.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. allow watch
5. session-server {session-tag[,...session-tag]}
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique ID of an already configured directory
Example: number. The tag number corresponds to a tag number
Router(config)# ephone-dn 24 created when this directory number was initially
configured.
Step 4 session-server Specifies which session managers are to monitor the
session-server-tag[,...session-server-tag] directory number being configured.
• session-server-tag—Unique ID session manager,
Example: configured in Unified CCX and automatically provided
Router(config-ephone-dn)# session-server to Cisco Unified CME. Range: 1 to 8.
1,2,3,4,6
Tip If you do not know the value for session-server-tag,
we recommend using 1.
Step 1 Use the show sip status registrar command to verify whether session manager and Cisco CRS route
points are registered.
Step 2 Use the show presence subscription summary command to verify whether Cisco CRS route points and
Unified CCX agent directory numbers are subscribed.
The following is sample output from the show presence subscription summary command. The first two
rows show the status for two route points. The next two are for logged in agent phones.
Router# show presence subscription summary
To re-create a session manager in Cisco Unified CME for Unified CCX, perform the following steps.
SUMMARY STEPS
1. enable
2. configure terminal
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register session-server Enters voice register session-server configuration mode to
session-server-tag enable and configure a session manager for an external
feature server, such as the Unified CCX application on a
Example: Cisco CRS system.
Router(config)# voice register session-server 1 • Range: 1 to 8.
• A single Cisco Unified CME can support multiple
session managers.
Step 4 register id name (Optional) Required only if the configuration from
Unified CCX is deleted or must be modified.
Example: • name—String for identifying Unified CCX. Can
Router(config-register-fs)# CRS1 contain 1 to 30 alphanumeric characters.
Step 5 keepalive seconds (Optional) Required only if the configuration from
Unified CCX is deleted or must be modified.
Example: • Keepalive duration for registration, in seconds, after
Router(config-register-fs)# keepalive 300 which the registration expires unless Unified CCX
reregisters before the registration expiry.
• Range: 60 to 3600. Default: 300.
Note Default in Unified CCX is 120.
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-fs)# end
Reconfiguring a Cisco CRS Route Point as a SIP Endpoint in Cisco Unified CME
To reconfigure a Cisco CRS route point as a SIP endpoint in Cisco Unified CME, perform the following
steps.
Prerequisites
• Directory numbers associated with Cisco CRS route points must be configured in
Cisco Unified CME. For configuration information for directory numbers associated with SIP
endpoints, see “Configuring Phones to Make Basic Calls” on page 147.
• Directory numbers associated with Cisco CRS route points must be enabled to be watched. For
configuration information, see “Configuring Presence Service” on page 797.
• The mode cme command must be enabled in Cisco Unified CME.
Restrictions
• Each Cisco CRS route point can be managed by only one session manager.
• Each session manager can manage more than one Cisco CRS route point.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. allow watch
6. refer target dial-peer
7. exit
8. voice register pool pool-tag
9. number tag dn dn-tag
10. session-server session-tag
11. codec codec-type [bytes]
12. dtmf-relay rtp-relay sip-notify
13. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config-register-global)# voice register
dn 1
Step 4 number number Defines a valid number for a directory number.
Example:
Router(config-register-dn)# number 2777
Step 5 allow watch Allows the phone line associated with this directory number
to be monitored by a watcher in a presence service.
Example:
Router(config-register-dn)# allow watch
Step 6 refer target dial-peer Enables watcher to handle SIP REFER message from this
directory number.
Example: • target dial-peer—Refer To portion of message is
Router(config-register-dn)# refer target based on address from dial peer for this directory
dial-peer number.
Step 7 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-dn)# exit
Step 8 voice register pool pool-tag Enters voice register pool configuration mode to set
device-specific parameters for a Cisco CRS route point.
Example: • A voice register pool in Unified CCX can contain up to
Router(config)# voice register pool 3 10 individual SIP endpoints. Subsequent pools are
created for additional SIP endpoints.
Step 9 number tag dn dn-tag Associates a directory number with the route point being
configured.
Example:
Router(config-register-pool)# number 1 dn 1
number 8001
allow watch
refer target dial-peer
!
voice register dn 3
session-server 1
number 8101
allow watch
refer target dial-peer
!
voice register dn 11
number 2011
name ep-sip-1-11
mwi
!
voice register dn 12
number 2012
name ep-sip-1-12
mwi
!
voice register dn 16
number 5016
name rp-sip-1-16
label SIP 511-5016
mwi
!
voice register dn 17
number 5017
name rp-sip-1-17
label SIP 511-5017
mwi
!
voice register dn 18
number 5018
name rp-sip-1-18
label SIP 511-5018
mwi
!
voice register pool 1
session-server 1
number 1 dn 1
number 2 dn 2
number 3 dn 3
dtmf-relay sip-notify
codec g711ulaw
!
voice register pool 11
id mac 1111.0711.2011
type 7970
number 1 dn 11
dtmf-relay rtp-nte
voice-class codec 1
username 5112011 password 5112011
!
voice register pool 12
id mac 1111.0711.2012
type 7960
number 1 dn 12
dtmf-relay rtp-nte
voice-class codec 1
username 5112012 password 5112012
!
voice register pool 16
id mac 0017.0EBC.1500
type 7961GE
number 1 dn 16
dtmf-relay rtp-nte
voice-class codec 1
username rp-sip-1-16 password pool16
!
voice register pool 17
id mac 0016.C7C5.0660
type 7971
number 1 dn 17
dtmf-relay rtp-nte
voice-class codec 1
username rp-sip-1-17 password pool17
!
voice register pool 18
id mac 0015.629E.825D
type 7971
number 1 dn 18
dtmf-relay rtp-nte
voice-class codec 1
username rp-sip-1-18 password pool18
!
!
!
!
!
!
!
controller T1 0/2/0
framing esf
clock source internal
linecode b8zs
pri-group timeslots 1-4,24
!
controller T1 0/2/1
framing esf
clock source internal
linecode b8zs
pri-group timeslots 1-4,24
!
controller T1 0/3/0
framing esf
clock source internal
linecode b8zs
ds0-group 0 timeslots 1-4 type e&m-immediate-start
!
controller T1 0/3/1
framing esf
clock source internal
linecode b8zs
ds0-group 0 timeslots 1-4 type e&m-immediate-start
vlan internal allocation policy ascending
!
!
!
!
interface GigabitEthernet0/0
ip address 209.165.201.1 255.255.255.224
duplex auto
speed auto
media-type rj45
!
interface GigabitEthernet0/1
ip address 192.0.2.254 255.255.255.0
duplex auto
speed auto
media-type rj45
!
interface Serial0/2/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
interface Serial0/2/1:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
interface Service-Engine1/0
ip unnumbered GigabitEthernet0/0
service-module ip address 209.165.202.129 255.255.255.224
service-module ip default-gateway 209.165.201.1
!
ip route 192.0.0.30 255.0.0.0 192.0.0.55
ip route 209.165.202.129 255.255.255.224 Service-Engine1/0
ip route 192.0.2.56 255.255.255.0 209.165.202.2
ip route 192.0.3.74 255.255.255.0 209.165.202.3
ip route 209.165.202.158 255.255.255.224 192.0.0.55
!
!
ip http server
ip http authentication local
ip http path flash:
!
!
ixi transport http
response size 64
no shutdown
request outstanding 1
!
ixi application cme
no shutdown
!
!
!
control-plane
!
!
!
voice-port 0/0/0
!
voice-port 0/0/1
!
voice-port 0/2/0:23
!
voice-port 0/3/0:0
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/2/1:23
!
voice-port 0/3/1:0
!
!
!
!
!
dial-peer voice 9000 voip
description ==> This is for internal calls to CUE
destination-pattern 9...
voice-class codec 1
session protocol sipv2
session target ipv4:209.165.202.129
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 9001 voip
description ==> This is for external calls to CUE
destination-pattern 5119...
voice-class codec 1
session protocol sipv2
session target ipv4:209.165.202.129
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 521 voip
destination-pattern 521....
voice-class codec 1
max-redirects 5
session protocol sipv2
session target ipv4:209.165.201.2
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 531 voip
destination-pattern 531....
voice-class codec 1
max-redirects 5
session protocol sipv2
session target ipv4:209.165.201.3
dtmf-relay rtp-nte sip-notify
!
!
presence
presence call-list
watcher all
allow subscribe
!
sip-ua
mwi-server ipv4:209.165.202.128 expires 3600 port 5060 transport udp
presence enable
!
!
telephony-service
no auto-reg-ephone
xml user axluser password axlpass 15
max-ephones 240
max-dn 720
ip source-address 192.0.2.254 port 2000
system message sb-sj3-3845-uut1
url services http://192.0.2.252:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
url authentication http:192.0.2.252:6293/ipphone/jsp/sciphonexml/IPAgentAuthenticate.jsp
cnf-file perphone
dialplan-pattern 1 511.... extension-length 4
voicemail 9001
max-conferences 8 gain -6
call-forward pattern .T
moh flash:music-on-hold.wav
multicast moh 239.10.10.1 port 2000
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Jun 18 2007 07:44:25
!
!
ephone-dn 1 dual-line
session-server 1
number 1001
name ag-1-1
allow watch
mwi sip
!
!
ephone-dn 2 dual-line
session-server 1
number 1002
name ag-1-2
allow watch
mwi sip
!
!
ephone-dn 3 dual-line
session-server 1
number 1003
name ag-1-3
allow watch
mwi sip
!
!
ephone-dn 4 dual-line
session-server 1
number 1004
name ag-1-4
allow watch
mwi sip
!
!
ephone-dn 5
session-server 1
number 1005
name ag-1-5
allow watch
mwi sip
!
!
ephone-dn 11 dual-line
number 3011
name ep-sccp-1-11
mwi sip
!
!
ephone-dn 12
number 3012
name ep-sccp-1-12
mwi sip
!
!
ephone-dn 16 dual-line
number 4016
label SCCP 511-4016
name rp-sccp-1-16
mwi sip
!
!
ephone-dn 17 dual-line
number 4017
label SCCP 511-4017
name rp-sccp-1-17
mwi sip
!
!
ephone-dn 18 dual-line
number 4018
label SCCP 511-4018
name rp-sccp-1-18
mwi sip
!
!
ephone-dn 19 dual-line
number 4019
label SCCP 511-4019
name rp-sccp-1-19
mwi sip
!
!
ephone-dn 20 dual-line
number 4020
label SCCP 511-4020
name rp-sccp-1-20
mwi sip
!
!
ephone-dn 21 dual-line
number 4021
label SCCP 511-4021
name rp-sccp-1-21
mwi sip
!
!
ephone-dn 22 dual-line
number 4022
label SCCP 511-4022
name rp-sccp-1-22
mwi sip
!
!
ephone 1
mac-address 1111.0711.1001
type 7970
keep-conference endcall
button 1:1
!
!
!
ephone 2
mac-address 1111.0711.1002
type 7970
keep-conference endcall
button 1:2
!
!
!
ephone 3
mac-address 1111.0711.1003
type 7970
keep-conference endcall
button 1:3
!
!
!
ephone 4
mac-address 1111.0711.1004
type 7970
keep-conference endcall
button 1:4
!
!
!
ephone 5
mac-address 1111.0711.1005
type 7970
keep-conference endcall
button 1:5
!
!
!
ephone 11
mac-address 1111.0711.3011
type 7970
keep-conference endcall
button 1:11
!
!
!
ephone 12
mac-address 1111.0711.3012
type 7960
keep-conference endcall
button 1:12
!
!
!
ephone 16
mac-address 0012.D916.5AD6
type 7960
keep-conference endcall
button 1:16
!
!
!
ephone 17
mac-address 0013.1AA6.7A9E
type 7960
keep-conference endcall
button 1:17
!
!
!
ephone 18
mac-address 0012.80F3.B013
type 7960
keep-conference endcall
button 1:18
!
!
!
ephone 19
mac-address 0013.1A1F.6282
type 7970
keep-conference endcall
button 1:19
!
!
!
ephone 20
mac-address 0013.195A.00D0
type 7970
keep-conference endcall
button 1:20
!
!
!
ephone 21
mac-address 0017.0EBC.147C
type 7961GE
keep-conference endcall
button 1:21
!
!
!
ephone 22
mac-address 0016.C7C5.0578
type 7971
keep-conference endcall
button 1:22
!
!
!
line con 0
exec-timeout 0 0
stopbits 1
line aux 0
stopbits 1
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
password lab
login
!
scheduler allocate 20000 1000
!
end
Where to Go Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME documentation roadmap
Cisco Unified Contact Center Express (Unified CCX) • Cisco Unified Contact Center Express documentation road map
Cisco Customer Response Solutions (CRS) • Cisco CRS Installation Guide
• Cisco CRS Administration Guide
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 56 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes SRST fallback support using Cisco Unified Communications Manager Express
(Cisco Unified CME).
Contents
• Prerequisites for SRST Fallback Support, page 943
• Restrictions for SRST Fallback Support, page 944
• Information About SRST Fallback Support, page 944
• How to Configure SRST Fallback Support, page 948
• Configuration Examples for SRST Fallback Support, page 953
• Additional References, page 955
• Feature Information for SRST Fallback Support, page 957
Before Fallback
1. Phones are configured as usual in Cisco Unified Communications Manager.
2. The IP address of the Cisco Unified CME router is registered as the SRST reference on the
Cisco Unified Communications Manager device pool.
3. SRST mode is enabled on the Cisco Unified CME router.
4. (Optional) Ephone-dns and features are prebuilt on the Cisco Unified CME router.
During Fallback
5. Phones that are enabled for fallback register to the default Cisco Unified CME router that has SRST
mode enabled. Each display-enabled IP phone displays the message that has been defined using the
system message command under telephony-service configuration mode. By default, this message is
“Cisco Unified CME.”
6. While the fallback phones are registering, the router in SRST mode initiates an interrogation of the
phones in order to learn their phone and extension configurations. The following information is
acquired or “learned” by the router:
– MAC address
– Number of lines or buttons
– Ephone-dn-to-button relationship
– Speed-dial numbers
7. The option defined with the srst mode auto-provision command determines whether
Cisco Unified CME adds the learned phone and extension information to its running configuration.
If the information is added, it appears in the output when you use the show running-config
command and is saved to NVRAM when you use the write command.
– Use the srst mode auto-provision none command to enable the Cisco Unified CME router to
provide SRST fallback services for Cisco Unified Communications Manager.
– If you use the srst mode auto-provision dn or srst mode auto-provision all commands, the
Cisco Unified CME router includes the phone configuration it learns from
Cisco Unified Communications Manager in its running configuration. If you then save the
configuration, the fallback phones are treated as locally configured phones on the
Cisco Unified CME-SRST router which could adversely impact the fallback behavior of those
phones.
8. While in fallback mode, Cisco Unified IP phones periodically attempt to reestablish a connection
with Cisco Unified Communications Manager every 120 seconds (default). To manually reestablish
a connection to Cisco Unified Communications Manager you can reboot the Cisco Unified IP
phone.
9. When a connection is reestablished with Cisco Unified Communications Manager, Cisco Unified IP
phones automatically cancel their registration with the Cisco Unified CME router in SRST mode.
However, if a WAN link is unstable, Cisco Unified IP phones can bounce between
Cisco Unified Communications Manager and the Cisco Unified CME router in SRST mode.
An IP phone connected to the Cisco Unified CME-SRST router over a WAN reconnects itself to
Cisco Unified Communications Manager as soon as it can establish a connection to
Cisco Unified Communications Manager over the WAN link. However, if the WAN link is unstable,
the IP phone switches back and forth between Cisco Unified CME-SRST and
Cisco Unified Communications Manager, causing temporary loss of phone service (no dial tone).
These reconnect attempts, known as WAN link flapping issues, continue until the IP phone
successfully reconnects itself back to Cisco Unified Communications Manager.
WAN link disruptions can be classified into two types: infrequent random outages that occur on an
otherwise stable WAN, and sporadic, frequent disruptions that last a few minutes.
To resolve WAN-link flapping issues between Cisco Unified Communications Manager and SRST,
Cisco Unified Communications Manager provides an enterprise parameter and a setting in the
Device Pool Configuration window called Connection Monitor Duration. (Depending on system
requirements, the administrator decides which parameter to use.) The value of the parameter is
delivered to the IP phone in the XML configuration file.
• Use the enterprise parameter to change the connection duration monitor value for all IP phones
in the Cisco Unified Communications Manager cluster. The default for the enterprise parameter
is 120 seconds.
• Use the Device Pool Configuration window to change the connection duration monitor value for
all IP phones in a specific device pool.
A Cisco Unified IP phone will not reestablish a connection with the primary
Cisco Unified Communications Manager at the central office if it is engaged in an active call.
Telephone Telephone
Fax
PSTN
V
Cisco Unified CME IP
router in SRST mode network
WAN
IP IP IP Cisco Unified IP phones disconnected
146571
PCs
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. srst mode auto-provision {all | dn | none}
5. srst dn line-mode {dual | single}
6. srst dn template template-tag
7. srst ephone template template-tag
8. srst ephone description string
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 srst mode auto-provision {all | dn | none} Enables SRST mode for a Cisco Unified CME router.
• all—Includes information for learned ephones and
Example: ephone-dns in the running configuration.
Router(config-telephony)# srst mode
auto-provision none
• dn—Includes information for learned ephone-dns in
the running configuration.
• none—Does not include information for learned
ephones or learned ephone-dns in the running
configuration. Use this keyword when you want
Cisco Unified CME to provide SRST fallback services
for Cisco Unified Communications Manager.
Step 5 srst dn line-mode {dual | single} (Optional) Specifies the line mode for ephone-dns in SRST
mode on a Cisco Unified CME router.
Example: • dual—SRST fallback ephone-dns will be dual-line
Router(config-telephony)# srst dn line-mode ephone-dns.
dual
• single—SRST fallback ephone-dns will be single-line
ephone-dns.
Note This command is used only when ephone-dns are
learned at the time of fallback. It is ignored when
you prebuild ephone-dn configurations.
Step 6 srst dn template template-tag (Optional) Specifies an ephone-dn template to be used in
SRST mode on a Cisco Unified CME router. The template
includes features that were specified when the template was
Example:
Router(config-telephony)# srst dn template 3
created. See “Configuring Templates for Fallback Support:
Example” on page 954.
• template-tag—Identifying number of an existing
ephone-dn template. Range is 1 to 15.
Step 7 srst ephone template template-tag (Optional) Specifies an ephone template to be used in SRST
mode on a Cisco Unified CME router.
Example: • template-tag—Identifying number of an existing
Router(config-telephony)# srst ephone ephone template. Range is 1 to 20.
template 5
Example:
Router(config-telephony)# end
Step 2 Use the show telephony-service ephone-dn command during fallback to review ephone-dn
configurations. Learned ephone-dns are noted by a line stating that they were learned during SRST
fallback.
Note Learned ephone-dns do not appear in the output for the show running-config command if the
none keyword is used in the srst mode auto-provision command.
ephone-dn 1 dual-line
number 4008
name 4008
description 4008
preference 0 secondary 9
huntstop
no huntstop channel
call-waiting beep
ephone-dn-template 8
This DN is learned from srst fallback ephones
Step 3 Use the show telephony-service ephone command during fallback to review ephone configurations.
Learned ephones are noted by a line stating that they were learned during SRST fallback.
Note Learned ephones do not appear in the output for the show running-config command if the none
keyword is used in the srst mode auto-provision command.
ephone 1
mac-address 0112.80B3.9C16
button 1:1
multicast-moh
ephone-template 5
Always send media packets to this router: No
Preferred codec: g711ulaw
user-locale JP
network-locale US
Description: "YOUR Description" : Oct 11 2005 09:58:27
This is a srst fallback phone
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. no service directed-pickup
5. create cnf-files
6. reset all
7. exit
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 no service directed-pickup (Optional) Disables directed call pickup and changes the
behavior of the PickUp soft key so that a user pressing it
invokes local group pickup rather than directed call pickup.
Example:
Router(telephony)# no service directed-pickup
This behavior is consistent with that of the PickUp soft key
in Cisco Unified Communications Manager.
Note For changes to the service-phone settings to be
effective, the Sep*.conf.xml file must be updated
with the create cnf-files command and the phone
units must rebooted with the reset command.
Step 5 create cnf-files Builds XML configuration files for Cisco Unified IP
phones.
Example:
Router(telephony)# create cnf-files
Step 6 reset all Resets all phones.
Example:
Router(telephony)# reset all
Step 7 exit Exits dial-peer configuration mode.
Example:
Router(telephony)# exit
ephone 2
mac-address 1002.CD64.A24A
type 7960
button 1:3
The following excerpt from the show running-config command displays the configuration of
ephone-dn 1 through ephone-dn 3. All three ephones are learned ephone-dns that are configured in
dual-line mode and use ephone-dn template 5, as specified in the telephony-service configuration mode
commands.
ephone-dn 1 dual-line
number 7001
description 7001
name 7001
ephone-dn-template 5
This DN is learned from srst fallback ephones
!
!
ephone-dn 2 dual-line
number 4005
name 4005
ephone-dn-template 5
This DN is learned from srst fallback ephones
!
!
ephone-dn 3 dual-line
number 4002
label 4002
name 4002
ephone-dn-template 5
This DN is learned from srst fallback ephones
ephone-dn 2
number 1102
name Register 2
ephone-dn 3
number 1103
name Register 3
ephone-dn 4
number 1104
name Register 4
ephone-dn 5
number 1105
name Register 5
ephone-dn 21
number 1121
name Park Slot 1
park-slot timeout 60 limit 3 recall alternate 1100
ephone-dn 22
number 1122
name Park Slot 2
park-slot timeout 60 limit 3 recall alternate 1100
ephone-template 5
fastdial 1 1101 name Front Register
fastdial 2 918005550111 Headquarters
softkeys idle Newcall Cfwdall Pickup
softkeys seized Endcall Cfwdall Pickup
softkeys alerting Endcall
softkeys connected Endcall Hold Park Trnsfer
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 57 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
This chapter describes the eXtensible Markup Language (XML) Application Programming Interface
(API) support available in Cisco Unified Communications Manager Express (Cisco Unified CME).
Contents
• Information About XML API, page 959
• How to Configure XML API, page 960
• Configuration Examples for XML API, page 965
• Where to Go Next, page 966
• Additional References, page 966
• Feature Information for XML API, page 968
Note The following Cisco IOS commands that were previously used with the XML interface are no longer
valid: log password, xmltest, xmlschema, and xmlthread.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip http server
4. ixi transport http
5. response size fragment- size
6. request outstanding number
7. request timeout seconds
8. no shutdown
9. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ip http server Enables the Cisco web browser user interface on the local
Cisco Unified CME router.
Example:
Router(config)# ip http server
Step 4 ixi transport http Specifies the XML transport method and enters
XML-transport configuration mode.
Example: • http—HTTP transport.
Router(config)# ixi transport http
Step 5 response size fragment-size Sets the response buffer size.
• fragment-size—Size of fragment in the response buffer,
Example: in kilobytes. Range is constrained by the transport type
Router(conf-xml-trans)# response size 8 and platform. See the CLI help for the valid range of
values.
Step 6 request outstanding number Sets the maximum number of outstanding requests allowed
for the transport type.
Example: • number—Number of requests. Range is constrained by
Router(conf-xml-trans)# request outstanding 2 the transport type and platform. See the CLI help for the
valid range of values.
Step 7 request timeout seconds Sets the number of seconds to wait, while processing a
request, before timing out.
Example: • seconds—Number of seconds. Range is 0 to 60.
Router(conf-xml-trans)# request timeout 30
Step 8 no shutdown Enables HTTP transport.
Example:
Router(conf-xml-trans)# no shutdown
Step 9 end Returns to privileged EXEC mode.
Example:
Router(config-xml-app)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. ixi application cme
4. response timeout {-1 | seconds}
5. no shutdown
6. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 ixi application cme Specifies the Cisco Unified CME application and enters
XML-application configuration mode.
Example:
Router(config)# ixi application cme
Step 4 response timeout {-1 | seconds} Sets a timeout for responding to the XML application and
overwrites the IXI transport level timeout.
Example: • -1—No application-specific timeout is specified. This
Router(config-xml-app) response timeout 30 is the default.
• seconds—Length of timeout, in seconds. Range is
0 to 60.
Step 5 no shutdown Enables XML communication with the application.
Example:
Router(conf-xml-app)# no shutdown
Step 6 end Returns to privileged EXEC mode.
Example:
Router(config-xml-app)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. xml user user-name password password privilege-level
5. end
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)# telephony-service
Step 4 xml user user-name password password Defines an authorized user.
privilege-level
• user-name—Username of the authorized user.
• password—Password to use for access.
Example:
Router(config-telephony)# xml user user23 • privilege-level—Level of access to Cisco IOS
password 3Rs92uzQ 15 commands to be granted to this user. Only the
commands with the same or a lower level can be
executed via XML. Range is 0 to 15.
Step 5 end Returns to privileged EXEC mode.
Example:
Router(config-telephony)# end
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. log table max-size number
5. log table retain-timer minutes
6. end
7. show fb-its-log
8. clear telephony-service xml-event-log
DETAILED STEPS
Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Router(config)#
Step 4 log table max-size number Sets the number of entries in the XML event table.
• number—Number of entries. Range is 0 to 1000.
Example: Default is 150.
Router(config-telephony)# log table max-size
100
Step 5 log table retain-timer minutes Sets the number of minutes to retain entries in the event
table before they are deleted.
Example: • minutes—Number of minutes. Range is 2 to 500.
Router(config-telephony)# log table Default is 15.
retain-timer 30
Example:
Router(config-telephony)# end
Step 7 show fb-its-log Displays the event logs.
Example:
Router# show fb-its-log
Step 8 clear telephony-service xml-event-log Clears XML event logs.
Example:
Router# clear telephony-service xml-event-log
Where to Go Next
For developer information on the XML API, see the XML Provisioning Guide for Cisco CME/SRST.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.
Note Table 58 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.
blf-speed-dial command 805, 807 call-forward b2bua noan command 367, 547, 713
BLF status 798 call-forward b2bua unreachable command 548
blocking call-forward busy command 525
automatic registration 129 call-forward busy command, overlaid ephone-dns 617
caller ID 467 call forwarding
call park 489 blocking local extension forwards 525
calls 467 H.450.3 standard 503
call transfer 503 selective 500
features 832 transcoding between G.726 and G.711 304
local directory 689 call-forward max-length command 525
bulk command 124 call-forward night-service command 525
bulk-loading speed-dial numbers 858 call-forward noan command 525
bulk registration call-forward noan command, overlaid ephone-dns 617
configuring 123 call-forward pattern command 524
disabling SIP proxy registration 172 call-forward system command 273
bulk-speed-dial list command 859 call hunt 565
bulk-speed-dial-prefix command 859 calling-number local command 534
busy call forwarding 500 callmon command 923
button command 188 call park
assigning dns to phones 161 alternate target 487
overlaid ephone-dns 618 blocking 489
button-layout command 898 call-park slots 486
dedicated slots 488
directed 486
C
examples 495
CA (certification authority) 392 monitoring call-park slots 486
cadence command 659 redirect 489
Callback soft key 830 reminder ring 487
call blocking timeout interval 487
based on date and time 467 call-park system redirect command 493
override 468 call pickup
call-coverage features 563 examples 621
caller-id block command 641 group numbers 566, 591
caller ID blocking 640 call routing, loopback 763
caller-id command 766 call setup, video 871
call-forward all command 525 call start slow command 872
call-forward b2bua all command 547 call transfer 550
call-forward b2bua busy command 366, 547 blocking 503
call-forward b2bua mailbox command 366, 547 consult transfer for direct station select 502
maximum sessions command 314, 662 prefix specification for SIP 380
max-pool command 136 mwi command 378, 380
max-redirect command 602 mwi-line command 188, 374, 375
max-subscription command 801 mwi prefix command 381
max-timeout command 599 mwi reg-e164 command 377
media encryption 387 mwi-server command 379
media messages 870 mwi stutter command 377
media path setup 871 mwi-type command 178, 376
media termination point, See MTP
meet-me conferencing 650
N
messages
debug 878 name command 160, 693, 699
flow control 871 network command 99
media 870 network-locale (ephone-template) command 297
MIBs 54 network-locale command 296
MIC (manufacture-installed certificate) 393 network locales
configuring 412 alternative 288
mode command 79, 136, 808 system-defined 288
mode ra command 408 user-defined 288
MOH (music on hold) network parameters 91
audio file to download 72 NewCall soft key 830
from a live feed 773 night service 579
from an audio file 774 examples 625
transcoding between G.726 and G.711 304 notification 579
moh (telephony-service) command 775 parameters 579
moh command 779 night-service bell (ephone) command 613
monitoring call-park slots 486 night-service bell (ephone-dn) command 612
monitor-line button 850 night-service call forwarding 500
MTP (media termination point) night-service code command 612
remote phones 156 night-service date command 610, 611
transcoding for video 869 night-service day command 610, 611
mtp command 197 night-service everyday command 610, 611
multicast moh command 775 night-service weekday command 612
multicast-moh command 776, 780 night-service weekend command 612
multi-party ad hoc conferencing 649 no-answer call forwarding 500
MWI no ephone command 82
configuring Subscribe notify 378 no-reg (ephone-hunt) command 600
configuring unsolicited notify 378 no-reg command 173
defining MWI outcall 377
telephony-service security parameters 414 private lines to Public Switched Telephone Network
(PSTN) 154
phone-key-size command 428
phone labels 895
profile-identifier register command 664
provision-tag command 232
phone number plan 268
phone-redirect-limit command 550
phones Q
analog 154
QCIF (one-quarter common intermediate format) 868
basic configuration 147
qsig decode command 536
configuration files 245
remote teleworker 155
phone screen R
custom background images 894
RA (registration authority) 392
header bar display 894
ready/not-ready status, hunt groups 577
system message display 895
Real-Time Transport Protocol, See RTP
phone-specific parameters for individual SIP phones 163
rebooting phones 257
phone user GUI access setup 348
Redial soft key 830
using CLI 349
refer-ood enable command 111
using GUI 348
refer target dial-peer command 929
pickup, See call pickup
regenerate command 422, 424
pickup-group command 592
register id command 927
PickUp soft key 830
register support, SIP 93
pilot command 598, 607
registrar command 97, 107, 124
PIN (personal identification number) 468
registrar server command 97, 924
pin command 477
registration, blocking automatic 129
PKI (Public Key Infrastructure) 391
registration, video-enabled endpoints 868
policy-list command 235
registration authority, See RA
port (CAPF-server) command 427
relay, DTMF 93
preference (ephone-dn) command 565, 587
reminder, call-park 487
preference (ephone-hunt) command 600
reminder, on-hold 820
preference (voice hunt group) command 608
remote phones 155
preference, dial-peer 565
request outstanding command 961
preference command 177, 589
request timeout command 961
preference command, overlaid ephone-dns 581, 616
reset (voice register global) command 262, 264
presence call-list command 801, 805, 807
reset (voice register pool) command 80
presence command 801, 809
resetting all SIP phones 261
presence enable command 801
resetting phones
presence service 797
description 257
present-call command 601
reset (ephone) command 259, 261
system administrator GUI access setup 343 transfer, See call transfer
system administrator security token, See SAST transfer max-length command 503, 528
TACACS authentication for HTTP server 341 translate command 276, 279