Cisco SES

Avaya Solution & Interoperability Test Lab
Configure Avaya Communication Manager and Avaya SIP Enablement Services for SIP Trunks with Cisco Unified CallManager Issue 1.0
Abstract
These Application Notes present a sample configuration for a network comprised of Avaya Communication Manager and Avaya SIP Enablement Services at the Main site, and Cisco Unified CallManager at the Branch site. SIP trunks are used to connect Avaya Communication Manager and Cisco Unified CallManager via Avaya SIP Enablement Services. All calls between the Main and Branch sites are carried over these SIP trunks. For the sample configuration, Avaya Communication Manager is running on the Avaya S8700 Media Servers with Avaya G650 Media Gateway. The results in these Application Notes should be applicable to other Avaya media servers and gateways.
TLT; Reviewed: SPOC 2/12/2007
Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved.
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1. Introduction
These Application Notes present a sample configuration for a network comprised of Avaya Communication Manager and Avaya SIP Enablement Services (SES) at the Main site, and Cisco Unified CallManager at the Branch site. The Avaya 4610SW SIP and H.323 Telephones at the Main site register locally to Avaya SES and Avaya Communication Manager respectively. The Cisco 7961 SIP and 7941 SCCP H.323 Telephones at the Branch site register locally to Cisco Unified Call Manager. SIP trunks are used to connect Avaya Communication Manager and Cisco Unified CallManager via Avaya SIP Enablement Services. All calls between the Main and Branch sites are carried over these SIP trunks. For the sample configuration, Avaya Communication Manager is running on the Avaya S8700 Media Servers with Avaya G650 Media Gateway. The results in these Application Notes should be applicable to other Avaya media servers and gateways.
A five digit Uniform Dial Plan (UDP) is used to facilitate dialing between the Main and Branch sites. Unique extension ranges are associated with Avaya Communication Manager at the Main site (33xxx), and Cisco Unified CallManager at the Branch site (56xxx). The detailed administration of the endpoint telephones is not the focus of these Application Notes and will not be described. For administration of endpoint telephones, refer to the appropriate documentation listed in Section 8. These Application Notes will focus on the configuration of the SIP trunks.
2. Equipment and Software Validated

The following equipment and software were used for the sample configuration provided: Equipment Avaya S8700 Media Servers Avaya G650 Media Gateway TN799DP C-LAN Circuit Pack TN2302AP IP Media Processor Avaya SIP Enablement Services Avaya 4610SW SIP Telephone Avaya 4610SW H.323 Telephone Avaya 6416D+ Digital Telephone Cisco Unified CallManager Cisco 7961 SIP Telephone Cisco 7941 SCCP H.323 Telephone Software Avaya Communication Manager 3.1.2, R013x.01.2.632.1 HW01 FW015 HW11 FW104 3.1, SES03.1-03.1.018.0 2.2.2 2.7 NA 5.0.4.2000-1 POS3-07-4-00 Ver 7.0 (2.0)
3. Configure Avaya Communication Manager

For successful interoperability with Cisco Unified CallManager, media shuffling has to be disabled. To maintain media shuffling amongst Avaya endpoints, and because calls to Cisco and Avaya SIP endpoints both require SIP trunks, separate SIP trunk groups along with separate signaling groups, network regions, and codec sets were created to provide the different treatments on media shuffling. In the test configuration, all Avaya endpoints (including the Avaya SIP endpoint) were configured to be in network region 1 with media shuffling enabled. Incoming SIP trunk calls from Avaya SIP endpoints will use network region 1 based on the IP network map configuration. Calls to Avaya SIP endpoints use network region 1 based on the Avaya SIP endpoints station mapping to a specific trunk group and signaling group. In the test configuration, network region 6 with media shuffling disabled was configured for Cisco endpoints. Incoming SIP trunk calls from Cisco endpoints will use network region 6 based on the signaling group configuration. Calls to Cisco endpoints will use network region 6 based on the AAR configuration, which selects a specific trunk group and signaling group. For this approach to enable shuffling among Avaya endpoints while preventing shuffling between Avaya and Cisco endpoints to be utilized, the signaling group whose far-end network region refers to the Cisco region must be a lower number than other signaling groups using the same near-end and far-end node names. When two SIP signaling groups use the same near-end and far-end node names, the lower numbered signaling group will be selected first by Avaya Communication Manager for incoming calls. It is also important to ensure sufficient trunks are allocated to the corresponding trunk group. Otherwise, in the case of insufficient trunks associated with the lower numbered signaling group, the higher numbered signaling group would be used. This could allow an incoming call from Cisco to be associated with the far-end region from the higher numbered signaling group that allows shuffling, resulting in no connected talk paths. If it is impractical to ensure that these conditions are met, then the same signaling group and trunk group can be used to reach both Avaya SIP and Cisco endpoints, with shuffling disabled on this signaling group. While not presented in these Application Notes, this simpler configuration where shuffling is disabled for all SIP calls has also been verified. The table below displays the configured network region, codec set, signaling group, trunk group associations and their purpose: Purpose For Cisco Endpoints For Avaya SIP Endpoints Region 6 1 Codec 6 1 Signaling 140 141 Trunk Intra-Region 140 141 Disabled Enabled Inter-Region Disabled Enabled
This section focuses on configuring the SIP trunks on Avaya Communication Manager to Avaya SES, which are used to reach the Avaya SIP endpoints and Cisco endpoints. In addition, this section highlights selected features that are required for the interoperability, and provides a sample routing using Automatic Alternate Routing (AAR). The configuration procedures include the following areas: Verify Avaya Communication Manager license Administer system parameters features Administer IP node names Administer IP interface and data module Administer IP codec sets and network regions Administer SIP trunk groups and signaling groups Administer SIP trunk group members and route patterns Administer location and public unknown numbering Administer uniform dial plan and AAR analysis Administer IP network map and station mapping
3.1. Verify Avaya Communication Manager License

Log into the System Access Terminal (SAT) to verify that the Avaya Communication Manager license has proper permissions for features illustrated in these Application Notes. Use the display system-parameters customer-options command and navigate to Page 2. Verify that there is sufficient remaining capacity for SIP trunks by comparing the Maximum Administered SIP Trunks field value with the corresponding value in the USED column. The difference between the two values needs to be greater than or equal to the desired number of simultaneous SIP trunk connections. When configuring the desired number of simultaneous SIP trunk connections, factor in the following: A call between an Avaya SIP endpoint and any Cisco endpoint uses two SIP trunks. A call between an Avaya non-SIP endpoint and any Cisco endpoint uses one SIP trunk. The license file installed on the system controls the maximum permitted. If there is insufficient capacity or a required feature is not enabled, contact an authorized Avaya sales representative to make the appropriate changes.
display system-parameters customer-options OPTIONAL FEATURES IP PORT CAPACITIES Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP eCons: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable H.323 Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: 5000 5000 0 0 0 10 0 0 50 Page 2 of 11
USED 151 13 0 0 0 0 0 0 20
3.2. Administer System Parameters Features

Use the change system-parameters features command to allow for trunk-to-trunk transfers. Submit the change. This feature is needed to be able to transfer an incoming call from the Branch back out to the Branch (incoming trunk to outgoing trunk), and to transfer an outgoing call to the Branch to another outgoing call to the Branch (outgoing trunk to outgoing trunk). For ease of interoperability testing, the Trunk-to-Trunk Transfer field was set to all to enable all trunkto-trunk transfers on a system wide basis. Note that this feature poses significant security risk, and must be used with caution. For alternatives, the trunk-to-trunk feature can be implemented on the Class Of Restriction or Class Of Service levels. Refer to the appropriate documentation in Section 8 for more details.
change system-parameters features Page 1 of FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 1 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? y Music/Tone on Hold: none Music (or Silence) on Transferred Trunk Calls? no DID/Tie/ISDN/SIP Intercept Treatment: attd Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred Automatic Circuit Assurance (ACA) Enabled? n 17
3.3. Administer IP Node Names

Use the change node-names ip command, and add entries for the C-LAN and SES server. In this case, clan2 and 192.45.100.70 are entered as Name and IP Address for the C-LAN, and MM-SES-server and 192.45.145.140 are entered as Name and IP Address for the SES server. The actual node names and IP addresses may vary. Submit these changes.
change node-names ip Name clan2 MM-SES-server IP NODE NAMES IP Address Name 192.45 .100.70 192.45 .145.140 Page 1 of 1
IP Address . . . . . .
3.4. Administer IP Interface and Data Module

Add the C-LAN to the system configuration using the add ip-interface n command, where n is an available slot number. Enter the C-LAN node name assigned from Section 3.3 into the Node Name field, and then the IP Address will be populated automatically. Enter proper values for the Subnet Mask and Gateway Address fields. In this case, 255.255.255.0 and 192.45.100.1 are used to correspond to the network configuration in these Application Notes. Set the Enable Ethernet Port field to y, and the Network Region field to the network region for Avaya endpoints from Section 3.5.1. Default values may be used in the remaining fields. Submit these changes.
add ip-interface 2a02 IP INTERFACES Page 1 of 1
Type: Slot: Code/Suffix: Node Name: IP Address: Subnet Mask: Gateway Address: Enable Ethernet Port? Network Region: VLAN:
C-LAN 02A02 TN799 D clan2 192.45 .100.70 255.255.255.0 192.45 .100.1 y 1 n
Link: Allow H.323 Endpoints? y Allow H.248 Gateways? y Gatekeeper Priority: 5
Target socket load and Warning level: 400 Receive Buffer TCP Window Size: 8320 ETHERNET OPTIONS Auto? y
Next, add a new data module using the add data-module n command, where n is an available extension. Enter the following values, and submit these changes. Name: Type: Port: Link: A descriptive name. ethernet Same slot number from the IP INTERFACES screen above and port 17. An available link number.
DATA MODULE Data Extension: Type: Port: Link: 24981 ethernet 02A0217 2 Name: CLAN 2A02 Data Module
add data-module 24981
Network uses 1's for Broadcast Addresses? y
3.5. Administer IP Codec Sets and Network Regions

Administer two IP codec sets, one to use for calls within the Main site (Avaya-Avaya), and the other to use for calls between the Main and Branch sites (Avaya-Cisco). Use the change ipcodec-set n command, where n is an existing codec set number to be used for the interoperability. Enter the desired audio codec type in the Audio Codec field. Retain the default values for the remaining fields and submit these changes. In the codec sets displayed below, codec set 1 was used for Avaya-Avaya calls, and codec set 6 was used for Avaya-Cisco calls. The actual codec set number and codec type may vary, and the same codec set number could have been used in the test configuration. Besides the G.711MU codec type, G.729B and G.729AB have also been verified to be interoperable with Cisco Unified CallManager via SIP trunks for the basic endpoint-to-endpoint call scenarios.
change ip-codec-set 1 IP Codec Set Codec Set: 1 Audio Codec 1: G.711MU 2: Silence Suppression n Frames Per Pkt 2 Packet Size(ms) 20 Page 1 of 2
change ip-codec-set 6 IP Codec Set Codec Set: 6 Audio Codec 1: G.711MU 2: Silence Suppression n Frames Per Pkt 2 Packet Size(ms) 20
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Administer two IP network regions, one to use for Avaya endpoints, and the other to use for Cisco endpoints. Use the change ip-network-region n command, where n is an existing network region number to be used for the interoperability.
3.5.1. IP Network Region for Avaya Endpoints

In the test configuration, network region 1 was used for Avaya endpoints. For the Location field, enter the location corresponding to the Main site from Section 3.8. For the Authoritative Domain field, enter the SIP domain name of the SES server from Section 4.1. Enter a descriptive Name. For the Codec Set field, enter the corresponding audio codec set number from the IP Codec Set screens for calls within Avaya. Enable the Intra-region IP-IP Direct Audio, Inter-region IP-IP Direct Audio, and IP Audio Hairpinning fields. These settings will enable direct media shuffling for Avaya-Avaya calls. Retain the default values for the remaining fields, and submit these changes.
change ip-network-region 1 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: mm.com Name: Avaya region MEDIA PARAMETERS Intra-region IP-IP Direct Audio: Codec Set: 1 Inter-region IP-IP Direct Audio: UDP Port Min: 2048 IP Audio Hairpinning? UDP Port Max: 65531 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? Call Control PHB Value: 34 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? Video PHB Value: 26 Page 1 of 19
yes yes y y y
Navigate to Page 3, and specify the appropriate codec set to use for calls between the Avaya network region 1 with the Cisco network region 6.
change ip-network-region 1 Inter Network Region Connection Management src rgn 1 1 1 1 1 1 1 dst codec direct Total rgn set WAN WAN-BW-limits 1 1 2 2 y :NoLimit 3 2 y :NoLimit 4 2 y :NoLimit 5 5 y :NoLimit 6 6 y :NoLimit 7 Video WAN-BW-limits :NoLimit :NoLimit :NoLimit :NoLimit :NoLimit Dyn CAC IGAR n n n n n Page 3 of 19
Intervening-regions
3.5.2. IP Network Region for Cisco Endpoints

In the test configuration, network region 6 was used for Cisco endpoints. For the Authoritative Domain field, enter the SIP domain name of the SES server from Section 4.1. Enter a descriptive Name. For the Codec Set field, enter the corresponding audio codec set number from the IP Codec Set screens for calls with Cisco. Disable the Inter-region IP-IP Direct Audio, Intra-region IP-IP Direct Audio, and IP Audio Hairpinning fields. These settings will disable direct media shuffling for Avaya-Cisco calls. Retain the default values for the remaining fields, and submit these changes. Note that the audio shuffling must be turned off, or else the calls with Cisco Unified CallManager will result in no connected audio paths.
change ip-network-region 6 IP NETWORK REGION Region: 6 Location: Authoritative Domain: mm.com Name: Cisco region MEDIA PARAMETERS Intra-region IP-IP Direct Audio: Codec Set: 6 Inter-region IP-IP Direct Audio: UDP Port Min: 2048 IP Audio Hairpinning? UDP Port Max: 65531 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? Call Control PHB Value: 34 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? Video PHB Value: 26 Page 1 of 19
no no n y y
Navigate to Page 3, and specify the appropriate codec set to use for calls between the Cisco network region 6 with the Avaya network region 1.
change ip-network-region 6 Inter Network Region Connection Management src rgn 6 6 dst codec direct Total rgn set WAN WAN-BW-limits 1 6 y :NoLimit 2 Video WAN-BW-limits :NoLimit Dyn CAC IGAR n Page 3 of 19
Intervening-regions
3.6. Administer SIP Trunk Groups and Signaling Groups

Administer two sets of SIP trunk groups and signaling groups, one set to reach the Avaya SIP endpoints, and the other to reach the Cisco endpoints.
3.6.1. SIP Trunk and Signaling Groups for Avaya SIP Endpoints
In the test configuration, trunk group 141 and signaling group 141 were used to reach the Avaya SIP endpoints. Use the add trunk-group n command, where n is an available trunk group number. Enter the following values for the specified fields, and retain the default values for the remaining fields. Group Type: Group Name: TAC: Service Type: sip A descriptive name. An available trunk access code. tie
Page TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 141 Group Type: SIP Trunk to Avaya COR: two-way Outgoing Display? n 0 tie Auth Code? sip 1 n CDR Reports: y TAC: 141 1 of 21
add trunk-group 141
TN: 1
Night Service: n Signaling Group: Number of Members: 0
Administer a SIP signaling group for the newly added trunk group to use for signaling. Use the add signaling-group n command, where n is an available signaling group number. Enter the following values for the specified fields, and retain the default values for all remaining fields. Submit these changes. Group Type: sip Transport Method: tls Near-end Node Name: C-LAN node name from Section 3.3. Far-end Node Name: SES server node name from Section 3.3. Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: Avaya network region number 1 from Section 3.5.1. Far-end Domain: SIP domain name of SES server from Section 4.1. DTMF over IP: rtp-payload
add signaling-group 141 SIGNALING GROUP Group Number: 141 Group Type: sip Transport Method: tls Far-end Node Name: MM-SES-server Far-end Listen Port: 5061 Far-end Network Region: 1 Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Session Establishment Timer(min): 120 Direct IP-IP Audio Connections? y IP Audio Hairpinning? Y Page 1 of 1
Near-end Node Name: clan2 Near-end Listen Port: 5061 Far-end Domain: mm.com
3.6.2. SIP Trunk and Signaling Groups for Cisco Endpoints

In the test configuration, trunk group 140 and signaling group 140 were used to reach the Cisco endpoints. Use the add trunk-group n command, where n is an available trunk group number. Enter the following values for the specified fields, and retain the default values for the remaining fields. Group Type: Group Name: TAC: Service Type: sip A descriptive name. An available trunk access code. tie
Page TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 140 Group Type: SIP Trunk to Cisco COR: two-way Outgoing Display? n 0 tie Auth Code? sip 1 n CDR Reports: y TAC: 144 1 of 21
add trunk-group 140
TN: 1
Night Service: n Signaling Group: Number of Members: 0
Navigate to Page 2, and enter 900 for the Preferred Minimum Session Refresh Interval (sec) field. This field sets the session refresh timer value. Avaya Communication Manager sends a session refresh request as a Re-INVITE or UPDATE after every timer interval to maintain ongoing sessions. When this value is less than 900, then Cisco Unified CallManager returns a 402 Session Timer Too Small response and results in further timer negotiations. Set this field to 900 to reduce unnecessary SIP messages.
add trunk-group 140 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 900 Page 2 of 21
Navigate to Page 3, and enter public for the Numbering Format field as shown below. Submit these changes.
display trunk-group 140 TRUNK FEATURES ACA Assignment? n Page Measured: none Maintenance Tests? y 3 of 21
Numbering Format: public Prepend '+' to Calling Number? n Replace Unavailable Numbers? n
Administer a SIP signaling group for the newly added trunk group to use for signaling. Use the add signaling-group n command, where n is an available signaling group number. Enter the following values for the specified fields, and retain the default values for all remaining fields. Submit these changes. Group Type: Transport Method: Near-end Node Name: Far-end Node Name: Near-end Listen Port: Far-end Listen Port: Far-end Network Region: Far-end Domain: DTMF over IP: sip tls C-LAN node name from Section 3.3. SES server node name from Section 3.3. 5061 5061 Cisco network region number 6 from Section 3.5.2. SIP domain name of SES server from Section 4.1. rtp-payload
Page SIGNALING GROUP Group Number: 140 Group Type: sip Transport Method: tls Far-end Node Name: MM-SES-server Far-end Listen Port: 5061 Far-end Network Region: 6 Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Session Establishment Timer(min): 120 Direct IP-IP Audio Connections? y IP Audio Hairpinning? Y 1 of 1
add signaling-group 140
Near-end Node Name: clan2 Near-end Listen Port: 5061 Far-end Domain: mm.com
3.7. Administer SIP Trunk Group Members and Route Patterns

Administer two sets of SIP trunk group members and route patterns to correspond to the two newly added SIP trunk groups. Use the change trunk-group n command, where n is a trunk group number added in Section 3.6. Enter the corresponding signaling group number from Section 3.6 into the Signaling Group field. Enter the desired number of trunk group members into the Number of Members field. Submit these changes.
change trunk-group 140 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 140 Group Type: SIP Trunk to Cisco COR: two-way Outgoing Display? n 0 tie Auth Code? sip 1 n CDR Reports: y TN: 1 TAC: 144 Night Service: n Signaling Group: 140 Number of Members: 10 Page 1 of 21
change trunk-group 141 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 140 Group Type: SIP Trunk to Avaya COR: two-way Outgoing Display? n 0 tie Auth Code? sip 1 n
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21
CDR Reports: y TN: 1 TAC: 141 Night Service:
n Signaling Group: 141 Number of Members: 10
Create two route patterns to use for the newly created SIP trunk groups. Use the change routepattern n command, where n is an available route pattern. Enter the following values for the specified fields, and retain the default values for the remaining fields. Submit these changes. Pattern Name: A descriptive name. Grp No: The trunk group number from above. FRL: Enter a level that allows access to this trunk, with 0 being least restrictive.
change route-pattern 140 Page Pattern Number: 140 Pattern Name: SES to Cisco SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted No Mrk Lmt List Del Digits Dgts 1: 140 0 2: 3: 4: 5: 6: BCC VALUE TSC CA-TSC 0 1 2 3 4 W Request 1: y y y y y n n ITC BCIE Service/Feature PARM 1 of 3
DCS/ QSIG Intw n n n n n n
IXC
user user user user user user
rest
No. Numbering LAR Dgts Format Subaddress none 1 of 3
change route-pattern 141 Page Pattern Number: 140 Pattern Name: SES to Avaya SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted No Mrk Lmt List Del Digits Dgts 1: 141 0 2: 3: 4: 5: 6: BCC VALUE TSC CA-TSC 0 1 2 3 4 W Request 1: y y y y y n n ITC BCIE Service/Feature PARM
DCS/ QSIG Intw n n n n n n
IXC
user user user user user user
rest
No. Numbering LAR Dgts Format Subaddress none
3.8. Administer Location and Public Unknown Numbering

Use the change locations command to assign the SIP route pattern for Avaya SIP endpoints to a location corresponding to the Main site. Add an entry for the Main site if does not exist already, enter the following values for the specified fields, and retain default values for the remaining fields. Submit these changes. Name: Timezone: Rule: Proxy Sel. Rte. Pat.: A descriptive name to denote the Main site. An appropriate timezone offset. An appropriate daylight savings rule. The Avaya route pattern number from Section 3.7.
Page LOCATIONS ARS Prefix 1 Required For 10-Digit NANP Calls? y Loc. Name No. 1: Main 2: Timezone Rule Offset + 00:00 0 NPA ARS FAC Attd FAC Prefix Proxy Sel. Rte. Pat. 141 1 of 16
change locations
Use the change public-unknown-numbering 0 command, to define the calling party number to be sent to Cisco Unified CallManager. Add an entry for the trunk group defined in Section 3.6.2 to reach Cisco endpoints. In the example shown below, all calls originating from a 5-digit extension beginning with 3 and routed to trunk group 140 will be sent as a 5-digit calling number. The calling party number will be sent to the far-end in the SIP From header. Submit these changes.
change public-unknown-numbering 0 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Ext Ext Trk Len Code Grp(s) Prefix Len Len Code Grp(s) 5 3 140 5 Page 1 of 2
CPN Prefix
Total CPN Len
3.9. Administer Uniform Dial Plan and AAR Analysis

This section provides a sample AAR routing used for routing calls with dialed digits 56xxx to Cisco Unified CallManager. Note that other methods of routing may be used. Use the change uniform-dialplan 0 command, and add an entry to specify use of AAR for routing of digits 56xxx. Enter the following values for the specified fields, and retain the default values for the remaining fields. Submit these changes. Matching Pattern: Len: Del: Net: Dialed prefix digits to match on, in this case 56. Length of the full dialed number. Number of digits to delete. aar
Page 1 of 2
change uniform-dialplan 0 UNIFORM DIAL PLAN TABLE
Percent Full: 0 Matching Insert Node Pattern Len Del Digits Net Conv Num 56 5 0 aar n Matching Insert Node Pattern Len Del Digits Net Conv Num n
Use the change aar analysis 0 command, and add an entry to specify how to route the calls to 56xxx. Enter the following values for the specified fields, and retain the default values for the remaining fields. Submit these changes. Dialed String: Total Min: Total Max: Route Pattern: Call Type: Dialed prefix digits to match on, in this case 56. Minimum number of digts. Maximum number of digits. The Cisco route pattern number from Section 3.7. aar
Page AAR DIGIT ANALYSIS TABLE Percent Full: Dialed String 56 Total Min Max 5 5 Route Pattern 140 Call Type aar Node Num ANI Reqd n 2 1 of 2
change aar analysis 0
3.10. Administer IP Network Map and Station Mapping

Use the change ip-network-map command to map the Avaya SIP endpoint IP address to the network region for Avaya endpoints from Section 3.5.1. This will enable the Avaya network region to be used for the Avaya SIP endpoint for incoming calls. A range of IP addresses may be configured when there is more than one Avaya SIP endpoint. Submit these changes.
change ip-network-map IP ADDRESS MAPPING Emergency Location Extension Page 1 of 32
From IP Address 192.45 .100.198 . . .
(To IP Address 192.45 .100.198 . . .
Subnet or Mask)
Region 1
VLAN n n
Use the change off-pbx-telephone station-mapping n command, where n is the extension number of the Avaya SIP endpoint. In the Trunk Selection field, enter the trunk group defined in Section 3.6.1 to reach Avaya SIP endpoints. This will enable the Avaya network region to be used for this Avaya SIP endpoint for outgoing calls. Submit these changes. Repeat this procedure for every Avaya SIP endpoint.
change off-pbx-telephone station-mapping 33001 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Extension 33001 Application OPS Dial Phone Number Prefix - 33001 Trunk Selection 141 Page 1 of 2
Configuration Set 2
4. Configure Avaya SIP Enablement Services

This section provides the procedures for configuring Avaya SIP Enablement Services. The procedures include the following areas: Obtain SIP domain Administer media server Administer host address map Administer host contact Administer trusted host
4.1. Obtain SIP Domain

Access the SES administration web interface by using the URL http://<ip-address>/admin in an Internet browser window, where <ip-address> is the IP address of the SES server. Note that the IP address for the SES server may vary, and in this case 192.45.145.140 is used, as administered in Section 3.3. Log in with the appropriate credentials and select the Launch Administration Web Interface option.
The Top screen is displayed, as shown below. If this is the initial setup of the SES server, then follow the Avaya SES documentation in Section 8 to administer the SIP domain and host. These Application Notes assume the SES server has already been configured with the proper domain and host information.
Select Server Configuration > System Properties from the left pane to display the Edit System Properties screen. Use the value in the SIP Domain field (in this case mm.com) for configuring the Authoritative Domain and Far-end Domain fields in Sections 3.5 and 3.6 respectively.
4.2. Administer Media Server

Select Media Servers > Add from the left pane to display the Add Media Server Interface screen. This screen associates a media server with a host. Enter the following values for the specified fields, and retain the default values for the remaining fields. Click on Add at the bottom of the screen to submit these changes. Media Server Interface Name: A descriptive name. Host: Select the IP address of the SES server from Section 4.1. SIP Trunk IP Address: Enter the C-LAN IP address from Section 3.3.
4.3. Administer Host Address Map

Select Hosts > List from the left pane to display the List Hosts screen. Click on the Map link.
In the List Host Address Map screen below, click on the Add Map In New Group link in the right pane.
The Add Host Address Map screen is displayed next. This screen is used to specify which calls are to be routed to Cisco Unified CallManager. For the Name field, enter a descriptive name to denote the routing. For the Pattern field, enter an appropriate syntax for address mapping. For the interoperability testing, a pattern of ^sip:56[0-9]{3} is used to match to any extensions in the range of 56000-56999 at the Branch site. Retain the check in Replace URI, and click Add.
4.4. Administer Host Contact

The List Host Address Map screen is displayed, and updated with the new address map. Click on Add Another Contact.
In the Add Host Contact screen, enter the contact sip:$(user)@<destination-IP-address> :5060;transport=udp, where the <destination-IP-address> is the IP address of the Cisco Unified CallManager from Section 5.1. Avaya SES will substitute $(user) with the user portion of the request URI before sending the message. Click on the Add button.
4.5. Administer Trusted Host

Administer Cisco Unified CallManager as a trusted host, so that the SIP messages from Cisco Unified CallManager will not be challenged by SES. To configure a trusted host, use the trustedhost a x n y c z command in the Linux shell of SES, where x is the IP address of the Cisco Unified CallManager from Section 5.1, y is the host name or IP address of the SES server from Section 4.1, and z is any desired comment.
craft@SES-DevCon1> trustedhost -a 192.45.130.105 -n 192.45.145.140 c Cisco 192.45.130.105 is added to trusted host list.
After configuring the trusted host, the user must go back to the SES administration web interface, and click on the Update link in the bottom left pane for all changes in Section 4 to take effect.
5. Configure Cisco Unified CallManager

The procedures for configuring Cisco Unified CallManager include the following areas: Log into CallManager Administer media resource group Administer media resource group list Administer SIP trunk security profile Administer SIP trunk Administer route pattern
Note that the default audio codec for all calls through Cisco Unified CallManager specifies G.711. Regions can be used to set up other audio codecs. Refer to the Cisco documentation in Section 8 for detailed procedures on how to configure regions.
5.1. Log Into CallManager

Access the Cisco Unified CallManager administration web interface by using the URL http://<ip-address> in an Internet browser window, where <ip-address> is the IP address of the Cisco Unified CallManager. Note that the IP address for the Cisco Unified CallManager may vary, and in this case 192.45.130.105 is used, and was configured as part of installation. Click on Cisco CallManager Administration at the bottom of the screen, and log in with appropriate credentials.
5.2. Administer Media Resource Group

The Cisco Unified CallManager Administration screen is displayed. Select Media Resources > Media Resource Group, as shown below.
The Find and List Media Resource Groups screen is displayed next. Click on Add New to add a new media resource group.
The Media Resource Group Configuration screen is displayed. Enter a descriptive Name and Description. Select the desired resources from the Available Media Resources section, and use the down arrow to move them to the Selected Media Resources section. For the interoperability testing, all available media resources were selected. Click Save.
5.3. Administer Media Resource Group List

Scroll to the top of the screen, and select Media Resources > Media Resource Group List, as shown below.
The Find and List Media Resource Group Lists screen is displayed next. Click on Add New to add a new media resource group list.
The Media Resource Group List Configuration screen is displayed. Enter a descriptive Name. Select the media resource group created in Section 5.2 from the Available Media Resource Groups section, and use the down arrow to move to the Selected Media Resource Groups section. Click Save.
5.4. Administer SIP Trunk Security Profile

Scroll to the top of the screen, and select System > Security Profile > SIP Trunk Security Profile, as shown below.
The Find and List SIP Trunk Security Profiles screen is displayed next. Click on Add New to add a new security profile for the SIP trunk to Avaya SES.
The SIP Trunk Security Profile Configuration screen is displayed. Enter a descriptive Name and Description. Select UDP from the Outgoing Transport Type field drop down list. Check the Accept Presence Subscription, Accept Out-of-Dialog REFER, Accept Unsolicited Notification, and Accept Replaces Header fields. Retain the default values in all remaining fields, and click Save.
5.5. Administer SIP Trunk

Scroll to the top of the screen, and select Device > Trunk, as shown below.
The Find and List Trunks screen is displayed. Click on Add New to add a new SIP trunk to Avaya SES.
The Trunk Configuration screen is displayed next. Select SIP Trunk from the Trunk Type field drop down list, and the value for the Device Protocol field will automatically change to SIP. Click on Next to proceed.
The Trunk Configuration screen is updated. In the Device Information section, enter the following values for the specified fields, and retain the default values for the remaining fields. Check the Media Termination Point Required field, which is required for all SIP trunk calls. Device Name: Description: Device Pool: Media Resource Group List: Media Termination Point Required: A unique identifier. A descriptive text. Default The media resource group list from Section 5.3. Check the field.
Scroll down to the Call Routing Information section. Check the Redirecting Diversion Header Delivery Inbound and Redirecting Diversion Header Delivery Outbound fields, to support call redirection information in the incoming and outgoing INVITE messages.
Scroll down to the SIP Information section. Enter the following values for the specified fields, and retain the default values for the remaining fields. Click Save. Destination Address: SIP Trunk Security Profile: SIP Profile: DTMF Signaling Method: IP address of Avaya SES server from Section 4.1. The SIP trunk security profile from Section 5.4. Standard SIP Profile RFC 2833
The message pop up box below is displayed, click OK.
Next, the screen is updated with additional buttons appearing at the bottom of the screen. Click Reset.
In the Device Reset dialog box, click on Reset, followed by Close.
5.6. Administer Route Pattern

Scroll to the top of the screen, and select Call Routing > Route/Hunt > Route Pattern, as shown below.
The Find and List Route Patterns screen is displayed. Click on Add New to add a new route pattern for the SIP trunk to use.
The Route Pattern Configuration screen is displayed. Enter a route pattern for the Route Pattern field, in this case 33XXX to denote the 5-digit numbering plan for extensions at the Main site. Enter a descriptive text for the Description field, and select the SIP trunk from Section 5.3 for the Gateway/Route List field drop down list. Uncheck the Provide Outside Dial Tone field. Retain the default values in the remaining fields, and scroll down to the bottom of the screen to click Save (not shown).
Click OK on the two subsequent pop up dialog boxes.
6. Verification Steps
This section provides the tests that can be performed on Avaya Communication Manager and Avaya SES, to verify proper configuration of Avaya Communication Manager, Avaya SES, and Cisco Unified CallManager.
6.1. Verify Avaya Communication Manager

Verify the status of the SIP trunk groups by using the status trunk n command, where n is the trunk group number administered in Section 3.6. Verify that all trunks are in the inservice/idle state as shown below.
status trunk 140 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy no no no no no no no no no no
0140/001 0140/002 0140/003 0140/004 0140/005 0140/006 0140/007 0140/008 0140/009 0140/010
T00351 T00352 T00353 T00354 T00355 T00366 T00367 T00368 T00369 T00370
in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle
status trunk 141 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy no no no no no no no no no no
0141/001 0141/002 0141/003 0141/004 0141/005 0141/006 0141/007 0141/008 0141/009 0141/010
T00356 T00357 T00358 T00359 T00360 T00361 T00362 T00363 T00364 T00365
in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle
Verify the status of the SIP signaling groups by using the status signaling-group n command, where n is the signaling group number administered in Section 3.6. Verify that the signaling group is in-service as indicated in the Group State field shown below.
status signaling-group 140 STATUS SIGNALING GROUP Group ID: Group Type: Signaling Type: Group State: 140 sip facility associated signaling in-service Active NCA-TSC Count: 0 Active CA-TSC Count: 0
status signaling-group 141 STATUS SIGNALING GROUP Group ID: Group Type: Signaling Type: Group State: 141 sip facility associated signaling in-service Active NCA-TSC Count: 0 Active CA-TSC Count: 0
Make a call within the Main site involving an Avaya SIP endpoint and an Avaya H.323 endpoint. Verify the status of connected SIP trunks by using the status trunk x/y, where x is the number of the SIP trunk group from Section 3.6.1 to reach Avaya SIP endpoints, and y is the member number of a connected trunk. Verify that the Service State is in-service/active, and that the IP addresses of the C-LAN and SES server are shown in the Signaling section. In addition, the Audio section shows the codec type and the IP addresses of the two Avaya endpoints. The Audio Connection Type displays ip-direct, indicating media shuffling.
status trunk 141/2 TRUNK STATUS Trunk Group/Member: 0141/002 Port: T00357 Signaling Group ID: Service State: in-service/active Maintenance Busy? no Page 1 of 2
Connected Ports: S00131
Port Signaling: 02A0217 G.711MU Audio: Video: Video Codec:
Near-end IP Addr : Port 192. 45.100. 70 : 5061 192. 45.100.133 : 12208
Far-end IP Addr : Port 192. 45.145.140 : 5061 192. 45.100.198 : 34008
Authentication Type: None Audio Connection Type: ip-direct
Make a call between the Main and the Branch site. Verify the status of connected SIP trunks by using the status trunk x/y, where x is the number of the SIP trunk group from Section 3.6.2 to reach Cisco endpoints, and y is the member number of a connected trunk. Verify that the Service State is in-service/active, and that the IP addresses of the C-LAN and SES server are shown in the Signaling section. In addition, the Audio section shows the codec type and the IP addresses of the Avaya IP Media Processor card and Cisco Unified CallManager. The Audio Connection Type displays ip-tdm, indicating no media shuffling.
status trunk 140/3 TRUNK STATUS Trunk Group/Member: 0140/003 Port: T00353 Signaling Group ID: Service State: in-service/active Maintenance Busy? no Page 1 of 3
Connected Ports: S00131
Port Signaling: 02A0217 G.711MU Audio: 02A0308 Video: Video Codec:
Near-end IP Addr : Port 192. 45.100. 70 : 5061 192. 45.100. 71 : 43972
Far-end IP Addr : Port 192. 45.145.140 : 5061 192. 45.130.105 : 25120
Authentication Type: None Audio Connection Type: ip-tdm
6.2. Verify Avaya SIP Enablement Services

From the Linux shell of Avaya SES, use the trustedhost L command to verify the IP address of the Cisco Unified Call Manager is listed as a trusted host.
craft@MM-SIP> trustedhost -L Third party trusted hosts. Trusted Host | CCS Host Name | Comment --------------------------+---------------------------+------------------------192.45.130.105 | 192.45.145.140 | Cisco
6.3. Verification Scenarios

Verification scenarios for the configuration described in these Application Notes included: Basic calls between various endpoints on the Main and Branch sites can be made in both directions via the SIP trunks using G.711MU, G.729B, and G.729AB. As mentioned previously, audio shuffling must be disabled. Proper display of the calling party name and number information were verified for all endpoints with the basic call scenario. The Avaya SIP endpoint displayed the calling party name, and all other endpoints displayed the calling party name and number. DTMF digits were verified by placing a call from Branch-to-Main that forwarded to a VDN on the Main site. The VDN played an announcement and prompted the caller to enter DTMF digits for alternate routing destination. The entered DTMF digits over the SIP trunks were verified to be recognized, and the call was successfully routed to the collected destination digits. This scenario was repeated with a call from Main-to-Branch that forwarded back to the same VDN on the Main site. Supplementary calling features were verified between various endpoints on the Main and Branch sites connected via SIP trunks. The feature scenarios involved additional endpoints on both the local and remote sites, such as performing an unattended transfer of the SIP trunk call to a local endpoint on the same site, and then repeating the scenario to transfer the SIP trunk call to a remote endpoint on the other site. The list of verified supplementary calling features include: o o o o o o Unattended transfer Attended transfer Hold/Unhold Consultative hold Call forwarding Conference
Proper display of only the calling party name information on all endpoints in both sites, when the Calling Number Block feature is activated from the sending site.
7. Conclusion
As illustrated in these Application Notes, Avaya Communication Manager can interoperate with Cisco Unified CallManager using SIP trunks via Avaya SES. The following is a list of interoperability items to note: Audio shuffling on Avaya Communication Manager, also referred to as direct IP-IP audio connection, must be disabled for calls between Avaya and Cisco endpoints. Media Termination Point on Cisco Unified CallManager must be checked, as it is required for all SIP trunk calls. The SIP session refresh interval on the Avaya SIP trunk group used to reach Cisco endpoints, can be set to 900 to avoid the 402 Session Timer Too Small response from Cisco Unified CallManager. The unattended trunk-to-trunk transfer scenario involving transfer of a Branch-Main call to another Branch-Main call in the alerting state does not always succeed. The workaround is to complete the transfer after the second Branch-Main call has been answered (i.e., attended transfer). The G.729B and G.729AB codecs are only interoperable for the basic endpoint-to-endpoint call scenarios, and not for more advanced scenarios such as transfers.
8. Additional References
This section references the product documentation relevant to these Application Notes. Administrator Guide for Avaya Communication Manager, Document 03-300509, Issue 2, February 2006, available at http://support.avaya.com. Installing and Administering SIP Enablement Services R3.1, Document ID 03-600768, Issue 1.4, February 2006, available at http://support.avaya.com. SIP Support in Release 3.1 of Avaya Communication Manager Running on the S8300, S8400, S8500 series, and S8700 series Media Server, Document 555-245-206, Issue 6, February 2006, available at http://support.avaya.com. Cisco Unified CallManager Administration Guide, Release 5.0(4), available at http://www.cisco.com.
2007 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please e-mail any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at interoplabnotes@list.avaya.com.

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Avaya Solution & Interoperability Test Lab

TLT; Reviewed: SPOC 2/12/2007

TLT; Reviewed: SPOC 2/12/2007

2. Equipment and Software Validated

TLT; Reviewed: SPOC 2/12/2007

3. Configure Avaya Communication Manager

TLT; Reviewed: SPOC 2/12/2007

3.1. Verify Avaya Communication Manager License

TLT; Reviewed: SPOC 2/12/2007

3.2. Administer System Parameters Features

3.3. Administer IP Node Names

TLT; Reviewed: SPOC 2/12/2007

3.4. Administer IP Interface and Data Module

C-LAN 02A02 TN799 D clan2 192.45 .100.70 255.255.255.0 192.45 .100.1 y 1 n

Link: Allow H.323 Endpoints? y Allow H.248 Gateways? y Gatekeeper Priority: 5

add data-module 24981

Network uses 1's for Broadcast Addresses? y

TLT; Reviewed: SPOC 2/12/2007

3.5. Administer IP Codec Sets and Network Regions

TLT; Reviewed: SPOC 2/12/2007

3.5.1. IP Network Region for Avaya Endpoints

TLT; Reviewed: SPOC 2/12/2007

3.5.2. IP Network Region for Cisco Endpoints

TLT; Reviewed: SPOC 2/12/2007

3.6. Administer SIP Trunk Groups and Signaling Groups

add trunk-group 141

Night Service: n Signaling Group: Number of Members: 0

TLT; Reviewed: SPOC 2/12/2007

TLT; Reviewed: SPOC 2/12/2007

3.6.2. SIP Trunk and Signaling Groups for Cisco Endpoints

add trunk-group 140

Night Service: n Signaling Group: Number of Members: 0

TLT; Reviewed: SPOC 2/12/2007

add signaling-group 140

TLT; Reviewed: SPOC 2/12/2007

3.7. Administer SIP Trunk Group Members and Route Patterns

CDR Reports: y TN: 1 TAC: 141 Night Service:

n Signaling Group: 141 Number of Members: 10

TLT; Reviewed: SPOC 2/12/2007

DCS/ QSIG Intw n n n n n n

user user user user user user

No. Numbering LAR Dgts Format Subaddress none 1 of 3

DCS/ QSIG Intw n n n n n n

user user user user user user

No. Numbering LAR Dgts Format Subaddress none

TLT; Reviewed: SPOC 2/12/2007

3.8. Administer Location and Public Unknown Numbering

Total CPN Len

TLT; Reviewed: SPOC 2/12/2007

3.9. Administer Uniform Dial Plan and AAR Analysis

change uniform-dialplan 0 UNIFORM DIAL PLAN TABLE

change aar analysis 0

TLT; Reviewed: SPOC 2/12/2007

3.10. Administer IP Network Map and Station Mapping

From IP Address 192.45 .100.198 . . .

(To IP Address 192.45 .100.198 . . .

TLT; Reviewed: SPOC 2/12/2007

4. Configure Avaya SIP Enablement Services

4.1. Obtain SIP Domain

TLT; Reviewed: SPOC 2/12/2007

TLT; Reviewed: SPOC 2/12/2007

TLT; Reviewed: SPOC 2/12/2007