Avaya 0B Configuration Note 88102 – Rev E (8/14)

Aura Messaging
Avaya CS1000
SIP Integration w/ Avaya Aura Session Manager

U Note : Integrating Aura Messaging

U

with multiple PBXs requires

special consideration
regarding Session Manager
Administration to ensure call
handling and MWI delivery. It
is advisable to consult with
your ATAC or Sales Engineer
representative.

Overview
This Configuration Note is intended for Avaya certified Aura Messaging
technicians/engineers who are familiar with Aura Messaging procedures and
terminology. It also assumes that you are Avaya certified or very familiar with the
features and functionality of the Avaya PBXs supported in this Configuration
Note and the SIP protocol.
Use this document in conjunction with Aura Messaging Installation Guide and the
appropriate Nortel PBX Guides mentioned throughout this Configuration Note.
Please read the entire document before attempting any configuration.

1.0 METHOD OF INTEGRATION

1B

SIP Trunks allows the Avaya The Session Initiation Protocol (SIP) integration provides connectivity
CS1000 PBX and the Avaya Aura with the Avaya PBX CS1000 over a Local Area Network (LAN). The
Messaging Server to communicate connectivity between the Avaya Aura Messaging Server and the PBX is
over a LAN. For multiple tandem achieved over an IP-connected SIP trunk via the Aura Session
CS1K connections behind Session Manager proxy. This integration passes call information and MWI using
Manger, you MUST too use SIP SIP packets.
trunk only to interconnect them.

Do not use PRI or H323 trunking.

Disclaimer: Configuration Notes are designed to be a general guide reflecting AVAYA Inc. experience configuring its systems. These
notes cannot anticipate every configuration possibility given the inherent variations in all hardware and software products. Please
understand that you may experience a problem not detailed in a Configuration Note. If so, please notify the Technical Service
Organization at (800) 876-2835, and if appropriate we will include it in our next revision. AVAYA Inc. accepts no responsibility for
errors or omissions contained herein.
 Avaya SIP Integration 2

Avaya Aura Messaging Server 2.0 AVAYA AURA MESSAGING SERVER REQUIREMENTS
2B

Requirements • Minimum releases required 1:

- Avaya Aura Messaging 6.x
p.- U
Release Note : U

Should features of the integration not

function optimally when integrated to a PBX
or Aura Messaging that may be operating
on an unsupported software release as 3.0 PBX HARDWARE REQUIREMENTS
defined Section 2.0 and 3.1, customers will Before performing the installation ensure the customer site has had an
need to upgrade their PBX and/or Aura Avaya Network Assessment and the customer has implemented the
Messaging to a supported software release.
recommendations.
PBX hardware requirements
• Avaya CS1000E CP+PM (Common Processor Pentium Mobile) Call
Server 7.5 (with Software as detailed below in Section 3.1)

3.1 PBX SOFTWARE REQUIREMENTS

Minimum Software 1 (see pg 2):

• Avaya CS1000E updated to the current DEPLIST

PBX/SESSION MANAGER • Avaya CS1000E with Release 7.5, Version 7.50.17

software requirements

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5B

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Avaya SIP Integration 3

3.2 SESSION MANAGER SOFTWARE/HARDWARE REQUIREMENTS

Minimum Supported Software and Hardware:
• Avaya Aura Session Manager 6.x
Hardware Required:
• Avaya S8xxx with SM100 card (acts as gateway to SM)
• Customer responsible for:
o Monitor, Keyboard, and Mouse
o Cat 5 Ethernet Cables
o Blank DVDs for burning ISO images if needed
Please refer to Installing and Administering Session Manager for more details.

3.3 CONNECTIVITY
6B

• Ethernet LAN connectivity – TCP/IP

3.4 CUSTOMER-PROVIDED EQUIPMENT

7B

• Wiring/equipment necessary to support the physical LAN (CAT 5

minimum)

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 Avaya SIP Integration 4

Supported integration features 4.0 SUPPORTED INTEGRATION FEATURES

[
] Items are supported

System Forward to Personal Greeting

All Calls [
]
Ring/no answer [
]
Busy [
]
Busy/No Answer [
]

Station Forward to Personal Greeting

All Calls [
]
Ring/no answer [
]
Busy [
]

Auto Attendant [
]
Call Me [
]
Direct Call [
]
External Call ID (ANI) [
]
Fax [
]
Find Me [
]
Internal Call ID [
]
Message Waiting Indication (MWI) [
]
Multiple Call Forward [
]
Multiple Greetings [
]
N+1 [
]
Outcalling [
]
Queuing [
]
Return to Operator [
]

U IMPORTANT : U PBX options or features not described in this Configuration

Note are not supported with this integration. To implement
options/features not described in this document, please
contact the Avaya Switch Integration product manager.

9B

Classic Nortel “extended” proprietary features such as ESN, DSC

and CDP topologies are not supported when connecting to Avaya
PBX equipment such - continued
as CM or on SM
next &
page – be disabled.
must
Multiple CS1K’s are supported in a “flat” hierarchy with no overlapping
extensions or mailboxes via SIP trunking between the CS1Ks.
DO NOT use H323 or PRI links between the tandem CS1Ks.

** IMPORTANT **
Encryption (TLS & SRTP) are currently not supported. Please
disable “msec” (media security) or leave to “Best Effort” in which
case AAM & CS1K will negotiate down to no security.

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 Avaya SIP Integration 5

5.0 CONFIGURING THE AVAYA CS1000E

Note: This Configuration Note assumes basic configuration of
telephones and SIP trunking to Session Manager has been
completed.
For information on basic configuration please refer to Communication
Server 1000E Installation and Commissioning. Release 7.5 Nortel
Doc#NN43041-310.
PBX Configuration The following tasks must be completed in the following order
when programming the PBX to integrate. PBX programming is
intended for certified PBX technicians/engineers.
U U

*Note: Avaya uses the

term “cover” while • Log in to CS1000E Element Manager
Nortel uses the term • Add a Distant Steering Code (DSC) for coverage and access
“forward.” to Aura Messaging
For purposes of this
• Configure phones to cover* to the Aura Messaging ‘pilot’
document they are one
extension
in the same.
• Log in to the Network Routing Service (NRS)
• Add a route for the Aura Messaging ‘pilot’ extension

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Avaya SIP Integration 6

5.1 Configuring the Avaya CS1000E

• This configuration uses the Avaya Aura Unified
Communication Management Server.
• Log in to the System Manager and choose UCM Services in
the Services column on the right.

• This will bring you to the Avaya Unified Communications

Management Elements page on the following page.

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 Avaya SIP Integration 7

• On the screen below, click on the Element Name that

corresponds to CS1000 in the Element Type column.

This section assumes

the SIP trunk between
Avaya Communication
Server 1000E and
Session Manager was
already configured

• Confirm Node and IP Addresses

o On the left side of the screen, expand System, and then
under that IP Network.
o Select Nodes: Servers, Media Cards.
o The IP Telephony Nodes page is now displayed as
shown below.
o In the Node ID column click on your specific ID to view its
details.
o In our example configuration, 1006 is our Node ID.

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Avaya SIP Integration 8

• Node Details
o The Node Details screen now appears as shown below.
o Make a note of both the Embedded LAN Call server IP
address and Telephony LAN Node Ipv4 address fields
outlined below. These values will be used to configure
other sections.

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Avaya SIP Integration 9

• Confirm Virtual D-Channel Configuration

The Avaya Communication Server 1000E Call Server uses a virtual D-
channel and associated Route and Trunks to communicate with the
Signaling Server. The following steps will guide you to ensure this
administration was completed.

• On the left side scroll down until you see Routes and Trunks.
• Expand that section and select D-Channels.
• The screen below shows the D-channels administered on the
sample configuration.
• In this configuration Channel 15 is our D-Channel, the Card Type
is DCIP. This denotes the Channel is a virtual (IP) D-channel.

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Avaya SIP Integration 10

• Confirm Routes and Trunks

o On the left side go to Routes and Trunks again.

o When expanded there is a sub-menu that has an option
with the same name Routes and Trunks (See screen on
previous page).
O Click on that sub-menu choice Routes and Trunks. The
screen as shown below will appear.
o This screen shows Route 15 configured with the Total
Trunks being 16. This means the system can handle 16
concurrent calls.
o To verify the configuration, select Edit.
o The screen on the next page is now displayed.

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Avaya SIP Integration 11

• Details of the and Trunks

Below are the Basic Configuration details of our virtual Route 15.
Ensure the following are set.

o Protocol ID for the route (PCID): SIP (SIP)

o Node ID of signaling server of this route (NODE): 1006
(This value matches the Node shown on the IP Telephony Screen on page 8)
o D channel number (DCH): 15
(This value matches the D-channel shown on the D-channels screen on page 10)

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 Avaya SIP Integration 12

• Route List Index and Distant Steering Code

You will now configure the routing of calls to Avaya Aura® Messaging.
Important:
You will first define the Route and then a Distant Steering Code.
The Rules, Routing,
and Distant Steering
Codes (DSC) shown • Create Route List Index
here are only
o On the left side expand Dialing and Numbering Plans.
examples.
o Select Electronic Switched Network.
Your Rules, Routing, o Select Route List Block (RLB) on the right side of the
and DSCs may be Electronic Switched Network (ESN) screen shown
different for your below.
customer network.

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Avaya SIP Integration 13

• The Route List Blocks screen is displayed.

o In the Please enter a route list index field, enter an
available route list index number
o Click to Add

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Avaya SIP Integration 14

• You will now see the Route List Block Screen as shown below.
• Under the Options section
o Select the Routing Number
(This was the Route ID from Page 11 and 12)
o Leave all remaining fields at their default values

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Avaya SIP Integration 15

• Add a Distant Steering Code (DSC)

o On the left of the of the ESN Screen (shown below),
expand Dialing and Numbering Plans
o Select Electronic Switched Network.
o In the Coordinated Dialing Plan (CDP) on the right side,
select Distant Steering Code (DSC)

o You will now see a Distant Steering Code List screen,

shown on the next page.

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Avaya SIP Integration 16

• Distant Steering Code List

o Select Add from the drop-down menu
o In the Please enter a distant steering code field, enter
the dialed prefix for external calls that will be routed over
the SIP trunk to Session Manager.
Note: In our sample configuration, our Distant Steering Code of 444 was
used as the Avaya Aura® Messaging Pilot Number was 444-5000,
and the Auto Attendant number 444-5001.
o Click to Add

• Enter the following for the fields in the screen below:

o Flexible Length number of digits: 7
(This is the number of digits in dialed numbers. In our configuration we used 7 digit numbers)
o Route List to be accessed for trunk steering code: 1
(This is the Route List Index we created in the Route List Blocks on page 14)

o Click Submit to save

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Avaya SIP Integration 17

• Private Network Identifier & MWI

To activate MWI, notify messages are sent from Avaya Aura® Messaging to
the Avaya CS1000E. To enable Avaya Communication Server 1000E to
receive SIP Notify messages from Avaya Aura® Messaging you need to
administer a Private Network Identifier for the system.
• On the left side of the screen (below) expand Customers.
o Select the customer (not shown in the screen below)
o On the right side select Feature Packages

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Avaya SIP Integration 18

o You will now see the Feature Packages page (not shown)
o Expand Integrated Services Digital Network
o In the Private network identifier field, enter 1*
*Note: A Private Network Identifier of 1 was used in our configuration. Your Private Network
Identifier may be different.

o Click Save

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Avaya SIP Integration 19

• Routes and Trunks

o On the left side expand Routes and Trunks
o Click on the sub-group Routes and Trunks

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Avaya SIP Integration 20

o Click on the Edit button on the right side that is associated

with the Route: 15.
(This was the Route ID we defined for the virtual D-Channel – see pages 11 and 12)

• The Basic Configuration section of Route 15 is now displayed.

o In the Private network identifier (PNI) field, enter 1
(This was the PNI you defined in the ISDN section – see page 19)

o Click Commit (not shown) to save changes.

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Avaya SIP Integration 21

• Verify MWI is enabled

o On the left side expand see Routes and Trunks.
o Select D-Channels.
o You will once again see the D-channels you administered
o In our configuration we used Channel 15 as our D-Channel
and the Card Type is DCIP, noting the Channel is a virtual
(IP) D-channel.
o Click the Edit button associated with this virtual D-Channel

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o On the next screen expand Basic Options (BSCOPT)

o Click Edit next to – Remote Capabilities

• On this next screen ensure Message waiting interworking

with DMS-100 (MWI) is enabled.

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Avaya SIP Integration 22

5.2 SUBSCRIBER ADMINISTRATION

Subscriber administration includes:
• Configure Phones to cover to the Aura Messaging ‘pilot’ extension
• Every Aura Messaging subscriber’s station/phone on the
CS1000E will need to be configured with the ‘pilot’ number of
4445000 so that busy and no-answer calls will route to Aura
Messaging. Although there are a number of tools that for
telephone administration on the CS1000E (i.e, Element Manager,
Telephony Manager, and the command-line overlay terminal) for this
document we will continue to use Element Manager to administer
the telephones.
• From the left-pane of Element Manager select Phones.
o Use the Search Criteria to select the station to edit a station.
The Value field is where you enter your station number.
.

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Avaya SIP Integration 23

In Features table, scroll to the Call Forward features

Set the following using the drop down menus

o CFHA – Call Forward/Hunt Override = Allowed
o CFXA – Call Forward External = Allowed

• Now scroll to the FDN – Flexible Call Forward No Ans DN

• In the adjacent field enter the Pilot Number, 4445000, for Avaya
Aura® Messaging

• Now scroll to the HTA – Hunting and set the following:

o HTA – Hunting = Allowed
o HUNT – Hunt DN – All Calls, or Internal Calls for CFTA =
Enter the Pilot Number, 4445000, for Avaya Aura® Messaging.

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Avaya SIP Integration 24

• Now scroll to the MWA – Message Waiting at Message Service

• Set it to Allowed

• Go to the Keys table

• Scroll down to Key No. 16
• Use the adjacent drop-down menu in the Key Type column and select
“MWK – Message Waiting
• In the Message Center DN field, enter the Pilot Number of 4445000
once again for Avaya Aura® Messaging.
• Click Save (not shown) to save your work.

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Avaya SIP Integration 25

• Ensure you SAVE your configuration by doing the following

o Expand Tools, then expand Backup and Restore
o Select Call Server, then Backup
o You will now see the Call Server Backup screen (shown below)
o On the right side select the Action Backup
o Then click Submit to save configuration

• The Backup process will take several minutes to complete.

• To see when the backup is complete, scroll to the bottom of the
page.

• This completes your configuration of your CS1000

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 Avaya SIP Integration 26

5.4 CONFIGURING THE AVAYA AURA SESSION MANAGER

Avaya Aura Session Manager routes calls between the Avaya
CS1000E and Avaya Aura Messaging.
Important:
Assumptions:
The domain names,
IP addresses, etc. • SIP entities for Avaya Communication Server 1000E and
provided in this Session Manager are defined.
document are only • Network connections are defined for:
examples. o Link between the Avaya System Manager and Avaya
Your domain names, Session Manager
IP addresses, etc. may o SIP trunk between Avaya CS1000E and Avaya Session
be different. Manager.

Most administration on Avaya Aura SM is done from the Network

Routing Policy screens accessed from the Routing section on the
left. For more complete programming information on Avaya Aura
Session Manager please refer to the appropriate documentation.

5.4.1 Define a SIP Domain

• Expand Elements, then Routing
• Select Domains from the left navigation menu.
• Click New (not shown).
• Enter the following values and use default values for remaining
fields.
o Name: avaya.com (our example Domain Name for the configuration. )
o Type: SIP
o Notes: <a brief description > (optional)
• Click Commit to save.

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Avaya SIP Integration 27

5.4.2 Define Location for Avaya Aura® Messaging

Locations identify logical and/or physical locations of SIP Entities. On
the left side of your screen:
• Expand Elements
• Expand Routing
• Select Locations
• Click New (not shown).

In the General section (shown below) enter the following:

Note: Leave all fields/values not noted set to their default
o Name: <a descriptive name for the location>
o Notes: <a brief description> [Optional]
• In the Location Pattern section
o Click Add and enter the following:
 IP Address Pattern: < the logical
pattern used to identify the location>
Note: In our example we entered 10.80.111.*

o Notes: <a brief description> [Optional]

• Click Commit to save.

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Avaya SIP Integration 28

5.4.3 Define a SIP Entity

You need to define a SIP Entity for Avaya Aura Messaging. On the left
side of your screen:
• Expand Elements
• Expand Routing
• Select SIP Entities
• Click New (not shown)
• In the General section (shown below) enter the following:
Note: Leave all fields/values not noted set to their default
o Name: <a descriptive name for the entity >
o Notes: <a brief description> [Optional]
o FQDN or IP Address: <Enter IP address of the Avaya Aura Messaging>
o Type: “Other”
o Notes: <a brief description> [Optional]
o Location: <select the Location defined for Avaya Aura Messaging. This is
the name you used for Location Details in the previous section. See 5.4.2>
• In the SIP Link Monitoring section (below) enter the following
o SIP Link Monitoring: Select Use Session Manager
Configuration
• Click Commit to save

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Avaya SIP Integration 29

5.4.4 Define a SIP Entity Link

You now need to define the SIP Entity Link between the Avaya Aura
Session Manager and Avaya Aura Messaging. On the left side of your
screen:
• Expand Elements
• Expand Routing
• Select Entity Links
• Click New (not shown) and enter the following:
o Name: <a descriptive name for the link>
o SIP Entity 1: ASM1 <this is the SIP Entity defined
for Avaya Session Manager>
o SIP Entity 2: Aura Messaging <this is the SIP
Entity defined for Avaya Aura Messaging. This is the
name you set in SIP Entity Details in the previous
section 5.4.3>
o Protocol: TCP <TCP was used in our sample
configuration , but TLS is also an option that could be
used. TLS provides security.>
o Port: 5060 <TCP uses port 5060; TLS uses port
5061>
o Trusted: check the box to set as a trusted
entity
o Notes: <a brief description> [Optional]
• Click Commit to save

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Avaya SIP Integration 30

5.4.5 Define Routing Policy for the Avaya Aura Messaging

Routing policies specifies the rules (policies) used to route between the
Avaya CS1000 and Avaya Aura Messaging. On the left side of your
screen:
• Expand Elements
• Expand Routing
• Select Routing Policies
• Click New (not shown)
• In the General section (shown below) enter the following:
Note: Leave all fields/values not noted set to their default
o Name: <a descriptive name/identifier for this Routing Policy>
o Disabled: <Leave unchecked>
o Notes: <a brief description> [Optional]
• In the SIP Entity as Destination section
o Click Select
o The SIP Entity List will now display (not shown).
o Select the Name of the SIP Entity used for Avaya Aura
Messaging. <this is the SIP Entity defined for Avaya Aura Messaging. This
is the name you set in SIP Entity Details in the previous section 5.4.2>
o Click Select.
o The Routing Policy Details is now displayed for Avaya Aura
Messaging (see screen below)
• Click Commit to save

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 Avaya SIP Integration 31

5.4.6 Define Routing Policy for the Avaya CS1000E

Now you repeat the steps as in the previous section 5.4.3. to define a
Routing Policy for the CS1000E. On the left side of your screen:
• Expand Elements
• Expand Routing
• Select Routing Policies
• Click New (not shown)
• In the General section (shown below) enter the following:
Note: Leave all fields/values not noted set to their default
o Name: <a descriptive name/identifier for this Routing Policy>
o Disabled: <Leave unchecked>
o Notes: <a brief description> [Optional]
• In the SIP Entity as Destination section
o Click Select
o The SIP Entity List will now display (not shown).
o Select the Name of the SIP Entity used for Avaya CS1000E
<this is the SIP Entity that we assumed was already defined for the Avaya
CS1000 that you are integrating with. If this Entity is not defined you will
need to add it following the same procedure when you added the Avaya
Aura Messaging as a SIP Entity. See section 5.4.2>
o Select the Name of the SIP Entity used for Avaya Aura
Messaging. <this is the SIP Entity defined for Avaya Aura Messaging.
This is the name you set in SIP Entity Details in the previous section 5.4.2>
o Click Select.
o The Routing Policy Details is now displayed for Avaya Aura
Messaging (see screen below)
• Click Commit to save

Note: The IP address

10.80.50.61 is the Node
TLAN IP Address in Node
Details in Section 5.1 (see
page 9 in this Config Note)

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Avaya SIP Integration 32

5.4.7 Define Dial Pattern for the stations

In our configuration we had defined two dial patterns for routing calls
between the Avaya CS1000E and Avaya Aura Messaging.
• 778 was defined for station numbering plans
• 4445 was defined for the Pilot Number
To define a Dial Pattern from the left side of your screen:
• Expand Elements
• Expand Routing
• Select Dial Patterns (not shown)
• Click New (not shown).
• In the General section, enter the following:
Note: Leave all fields/values not noted set to their default
o Pattern: 778 <this is the dial pattern for the stations>
o Min: 7 < this is the minimum number digits required to dial>
o Max: <Enter the maximum number digits that may be dialed>
o SIP Domain: ALL <select the specific SIP Domain from drop-down menu or
ALL if Session Manager should accept incoming calls from all SIP Domains>
o Notes: <a brief description> [Optional]
• In the Originating Locations and Routing Policies section
o Click Add
o The Originating Locations and Routing Policy List is now
displayed (not shown).
o In the Originating Locations table select ALL
o Notes: <a brief description of the Dial Pattern>
[Optional]
o In the Routing Policies table, select the Routing Policy defined
for Avaya Communication Server 1000E in Section 5.4.6
o Click Select to save your changes and return to Dial Pattern
Details page.
• Click Commit to save

o
o

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Avaya SIP Integration 33

5.4.8 Define Dial Pattern for the Pilot Number

Here we repeat the steps as in section 5.4.7 to define the dial pattern to
route calls to the Pilot Number for Avaya Aura Messaging.
To define this Dial Pattern from the left side of your screen:
• Expand Elements
• Expand Routing
• Select Dial Patterns (not shown)
• Click New (not shown).
• In the General section, enter the following:
Note: Leave all fields/values not noted set to their default
o Pattern: 4445 <this is the dial pattern for the Pilot Number>
o Min: 7 < this is the minimum number digits required to dial>
o Max: <Enter the maximum number digits that may be dialed>
o SIP Domain: ALL <select the specific SIP Domain from drop-down menu or
ALL if Session Manager should accept incoming calls from all SIP Domains>
o Notes: <a brief description> [Optional]
• In the Originating Locations and Routing Policies section
o Click Add
o The Originating Locations and Routing Policy List is now
displayed (not shown).
o In the Originating Locations table select ALL
o Notes: <a brief description of the Dial Pattern>
[Optional]
o In the Routing Policies table, select the Routing Policy defined
for Avaya Communication Server 1000E in Section 5.4.6
o Click Select to save your changes and return to Dial Pattern
Details page.
• Click Commit to save

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 Avaya SIP Integration 34

6.0 CONFIGURING THE AURA MESSAGING SERVER

The procedures to complete the integration of Avaya Aura
Messaging will require the following Administration:
• Adding Site (Avaya Aura Messaging Server)
• Configure the Telephony Integration
Configuring the Aura
Messaging Server 6.0.1 Administer Messaging
• Select Administration
• Select Messaging

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 Avaya SIP Integration 35

6.0.2 Add Site (Avaya Messaging)

On the left side of the screen under Messaging System (Storage):
• Select Sites
The Sites screen will display (shown below). On the right
• In the Sites section under Main Properties, enter the following:
o Name: Avaya Messaging <a descriptive name/identifier for this site>
o Messaging access number (external): 4445000 <the Pilot number
Note: for the site>
In our configuration we used - o Messaging access number (internal): 4445000 <the Pilot number
444-5000 as our Pilot # for the site>
- and - o Extension Length: 7 <the number of digits in the station numbers>
444-5001 as our Auto Attendant # o Mailbox Length: 7 <the number of digits in the mailbox numbers>
• In the Sites section under Auto Attendant:
The Pilot and Auto Attendant
Numbers in your configuration o Auto Attendant: Enable <click on the radio button to enable>
may be different. o Auto Attendant Pilot Number: 444-5001 <the Auto Attendant
number for the site>
o Click Save to save your site definition

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
Avaya SIP Integration 36

6.0.3 Administering the Telephony Integration

• In the Administration / Messaging menu on the left side, under
Telephony Settings (Application)
o Select Telephony Integration
The Telephony Integration Screen will display (shown below).
There are two sections to this screen that will be used to complete
the Telephony Integration of Avaya Aura Messaging. They are:
o Basic Configuration
o SIP Specific Integration
These sections will be detailed in the next pages/sections

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 Avaya SIP Integration 37

Note: 6.0.3.1 Basic Configuration

Please note, the settings The BASIC CONFIGURATION section of the Telephony Integration
shown in Basic and SIP Screen is shown below. Please refer to the configuration details
Specific Configuration are shown below the screen to administer your site.
for example only. Your
settings may be different.

BASIC CONFIGURATION (Parameters/Settings):

• Switch Number = 1 <always set to 1 unless Avaya Support directs

otherwise>
• Extension Length = 7 <extension length up to 10 digits>. The
extension length of 7 in the screen matches the dial plan of the media
server.
• Switch Integration Type = SIP
• IP Address Version = Ipv4
• Quality of Service = A Value of 0 to 63 may be used.

The Call Control PHB and Audio PHB set the QOS levels for call
control and audio messages on networks that support this feature.

These values must match the corresponding numbers on page 1 of

the IP Network Region screen (see IP Network Region screen in
Section 5.1 in this CN) under the DIFFSERV/TOS PARAMETERS.
If numbers do not match then call failures may result.

• UDP Port Range – Start = 8000 / End = 8410

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
 Avaya SIP Integration 38

Note: 6.0.3.2 SIP Specific Configuration

Please note, the settings The SIP SPECIFIC CONFIGURATION section of the Telephony
shown in Basic and SIP Integration Screen is shown below.
Specific Configuration are
for example only. Your
settings may be different.

SIP SPECIFIC CONFIGURATION (Parameters/Settings):

• Transport Method = TCP or TLS*. <This is the transport method

used for SIP signaling and must match the transport method administered on
the switch>
** IMPORTANT ** • Far-end Connections = 1. <This is the number of far-end connections
Encryption (TLS & SRTP) are to administer>
currently not supported. Please
disable “msec” (media security) or • Connection 1 = 10.80.111.102 <This is the IP Address of the Aura
leave to “Best Effort” in which case Session Manager>
AAM & CS1K will negotiate down to
no security. o Port number = 5060 (5060 is for TCP ; 5061 for TLS)
• Messaging Address = 10.80.111.102 <This is the IP address of Aura
Messaging as defined in section 5.4.3>
o Port number = 5060 (5060 is for TCP ; 5061 for TLS)
• SIP Domain = <domain name> <This is the Domain Name we used for
our configuration, see Section 5.4>
• Messaging Ports –
o Call Answering Ports = 2 or more. <The number of call answering
ports configured on the system. This could be less than or equal to the
maximum number of ports available
o Maximum = xxx <The maximum number of ports that may be configured
as Call Answering ports>.

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
 Avaya SIP Integration 39

o Transfer Ports = xx <This field is read only and shows the ports available
for transfer ports. This is calculated as the difference between the number of
trunks and call answer ports>
• Switch Trunks = xxx <Must match the number of trunks configured for the
messaging on the switch. If multiple signal groups are administered, this number is the
sum of all trunks in all groups>
** IMPORTANT ** • Media Encryption** = None
Encryption (TLS & SRTP) are
currently not supported. Please srtp-aescm128-hac80
disable “msec” (media security) or srtp-aescm128-hmac32
leave to “Best Effort” in which case
AAM & CS1K will negotiate down to
Chose the encryption (one of the 3 choices noted above) that matches
no security. what is administered in the IP Codec Set used for this integration.

Click Save to save all changes.

Once this is done, the Switch Link Administration screen will be

displayed notifying the user if a restart is required (sometimes a
restart is not needed) for changes to take effect.

- continued on next page –

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
Avaya SIP Integration 40

p.4- MESSAGING CAPACITY CALCULATOR

The button labeled “Show Capacity Calculator” just below the SIP
SPECIFIC CONFIGURATION section is a tool that can be used to
determine the number of call answer ports needed. If you click on
Show Capacity Calculator the following screen appears.

HOW TO USE THE CALCULATOR

Traffic Load – Chose a traffic load profile that suits your needs.
Table A (below) is a guideline to help determine traffic load.

Voice Port Usage in

Number of Voice Messages
Traffic Load Minutes
(per subscriber per day)
(per subscriber per day)
Light 2 1.5
Medium 4 3
Heavy 6 4.5
Very Heavy 8 6
Extremely Heavy 10 7.5
Table A. Traffic Load Guide
• Minimum Number of Voice Ports - <Enter the number of call answer ports
(must be at least 2) >
• Click on the Calculate Mailboxes button to display the number of
mailboxes (as recommended by Avaya)

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
Avaya SIP Integration 41

1
• Maximum Number of Mailboxes – Enter the number of mailboxes
(must be at least 2).
• Click on the Calculate Ports button to display the number of call
answer ports <as recommended by Avaya>
1
Maximum number of mailboxes is determined by your license.

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
 Avaya SIP Integration 42

1 8.0 CONSIDERATIONS / ALTERNATIVES

** IMPORTANT ** 8.1 SIP integrations may not be reliable for TTY/TDD if the IP
Encryption (TLS & SRTP) are network is unable to support uncompressed audio with no packet loss.
currently not supported. Please
disable “msec” (media security) or
For this reason Avaya does not support TTY/TDD with this SIP
leave to “Best Effort” in which case integration.
AAM & CS1K will negotiate down to 8.2 In reference to supported “transport CODECs”, AAM supports
no security. only G.711. Ensure the far end SIP end point (SIP gateway or
SIP PBX) is set accordingly. Failure may result in undesirable
or what’s perceived to be a non-working or dysfunctional AAM.
G.711 is the front-ended transport CODEC, AAM’s back-end
storage allows for both GSM and G.711 CODECs to the
message store. This latter switch setting is found within the SMI
of “System Parameters”.
8.3 If your integration is set to use TLS as the transport
method/link type and calls are not completing but they do
complete using TCP, then the cause may be a license issue.
8.4 Avaya Aura Messaging currently does not support E.164
formatted numbering or any mailbox or extension number
exceeding 10-digits. You may need to add an Adaptation Rule to
Session Manager to add/subtract digits or the Routing Pattern
entry to handle this.
8.5 If you are using Outlook and attempt to Play a message on a
phone that requires an outside trunk and the call get
rejected/fails, check to see if service provide is blocking calls
with names.
8.6 In a network consisting of an Avaya CM and CS1000 with a
Session Manager, if a call originates from a station on CM to a
station on the CS1000, and subsequently gets transferred to
another station on the same CS1000 (for example in a zero out
scenario) the caller may experience no talk path. The
workaround for this issue is to disable a feature in the CM SIP
trunk-group called Network Call Redirection (NCR).
8.7 It’s important to note special CS1K phone terminal PBX programming
is warranted when having multiple DN extensions to a singular
physical phone (i.e. main is 7000 (secretary) with execs on 7001, 7002
and 7003).
Real CLID values must be put into the configuration of the additional
keys on the target set configuration. If you configure a "D" instead of
a real CLID entry, it uses the key with CLID entry before the "D" as

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
Avaya SIP Integration 43

the digits that go to Aura Messaging. For example if key 0 is ext 7000
with a CLID 0 and key 1 is ext 7001 and has CLID D, 7001 will go to
mailbox 7000. If you change 7001 to have a CLID 0 or any other
number it will go into its own mailbox.
For more detailed information, see CS1K documentation and search
for keyword “KEY”. It’s switch syntax is “KEY xx aaa yyyy (cccc or
D) zz..z”.

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
Avaya SIP Integration 44

CHANGE HISTORY
Issue
Version Reason for Change
Date
A 1/27/2012 Moving document to GA status. No content changes.
Added a special note to section 8.0 to address multiple DNs
B 4/25/2012
(extensions) to one physical phone. Configuration clarification concern.
C 5/9/2012 Clarification under Section 8 regarding CODECs.
D 6/25/2012 Removed DRAFT note.
E 8/26/2014 Removed need for CS1K related Premium Feature set.

©2011 AVAYA Inc. All rights reserved. All trademarks identified by the ®, SM and TM are registered trademarks,
service marks or trademarks respectively. All other trademarks are properties of their respective owners. The
above information is based on knowledge available at the time of publication and is subject to change without
notice. Printed in U.S.A.

AVAYA Inc.
4655 Great America Parkway
Santa Clara CA 95054
+1-866-Go-Avaya
From Outside the US: +1 (908) 953-6000
U http://www.avaya.com

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1
Avaya SIP Integration 45

ADDENDUM FOR AUDIOCODES GATEWAY INTEGRATIONS

This section contains information regarding Issues and Solutions found

with AudioCodes Gateways integrations.
Note for AAM: Only AudioCodes firmware version 5.80A.xxx.xxx is supported.

1. Issue: FIND ME: On a Find Me call when the called party answers they hear four DTMF
digits (A, B, C, D) are played followed by about 1 second of silence,
followed by the normal prompt with the first little bit missing).
SOLUTION: In the AudioCodes .ini file Add the RxDTMFHangOverTime parameter
with a value of 100 instead of the default value of 1000ms.
2. Issue: DTMF - User presses the # key in a recording which is translated to a slight
"bleep" when the recording is listened to.
SOLUTION: You can reduce the length of the DTMF chirp using a procedure for
changing the recognition of DTMF in the AudioCodes. Please contact
Integrations Support for this information.
3. Issue: FAC - Transfer to Voice Mail is a feature that is currently NOT SUPPORTED when
using AudioCodes Gateways. A solution is currently under investigation.
4. Issue: Transfer/FINDME Fails - Calls originating through one Mediant Gateway to AAM, that have
a new independent call established from the AAM through Mediant B will ring
the end user but when call is answered user hears a tone and call is
disconnected and a SIP 481 error is generated in the logs. Call is split and
cannot be bridged as GWs do not know each has a leg of the same call.
SOLUTION: Use one Gateway. Multiple gateways are currently not supported
5. Issue: Beep tone - A beep tone is heard when on a transfer just before the Personal Greeting is
played. On a RNA no tone is heard.
SOLUTION: This occurs because AAM sends an sdp with (audio) “a=inactive.” This then
causes the Mediant gateway to play a HELP_TONE because it assumes that
MoH (Music on Hold) will have to be played locally since there is no audio
stream expected (a=inactive). The only way around this is to remove the tone
from the CPT file in the Gateway. A CPT with this tone removed is available
from Integrations Support.
6. Issue: E1 calls fail on upper half of span - If calls on E1 channels above 16 (the D-Channel for an
E-1) have no talk path (dead air) it may be a setting in the AudioCodes
Gateway causing it.
SOLUTION: In the AudioCodes ini file, check the ISDNGeneralCCBehavior parameter to
see if it is set to 32. If so change it to 0, which is the default value. Then
reload/burn the INI and calls should complete properly.

The above information is provided by AVAYA Inc. as a guideline. See disclaimer on page 1