Configuring SIP Trunks among Cisco Unified

WDC Reviewed: SPOC 12/10/2010

Solution Interoperability Lab Application Notes ©2010 Avaya Inc. All Rights Reserved.

1 of 71 SM6CUCM8CMESSIP

Avaya Solution Interoperability Lab

Configuring SIP Trunks among Cisco Unified Communications Manager 8.0.3, Avaya Aura™ Session Manager 6.0 and Avaya Aura™ Communication Manager 6.0 installed as an Evolution Server – Issue 1.0

Abstract

These Application Notes describe a sample configuration of a network that uses SIP trunks between Cisco Unified Communications Manager, Avaya Aura™ Session Manager and Avaya Aura™ Communication Manager installed as an Evolution Server.

Avaya Aura™ Session Manager provides SIP proxy/routing functionality, routing SIP sessions across a TCP/IP network with centralized routing policies and registrations for Avaya SIP endpoints.

Avaya Aura™ Communication Manager operates as an Evolution Server for the SIP endpoints which communicates with Avaya Aura™ Session Manager over SIP trunks.

Cisco Unified Communication Manager allows SIP Trunk interconnectivity with other PBX systems and supports Cisco IP Phones supporting SIP and SCCP protocols.

This Application Note provides information for the setup, configuration, and verification of the call flows tested on this solution.




Table of Contents

1. INTRODUCTION................................................................................................................. 4

1.1. EQUIPMENT AND SOFTWARE VALIDATED .............................................................................................................. 5

2. CONFIGURE AVAYA AURA™ COMMUNICATION MANAGER EVOLUTION SERVER ............................................................................................................................... 6

2.1. VERIFY SYSTEM CAPABILITIES AND LICENSING ........................................................................................................ 6 2.1.1. SIP Trunk Capacity Check ....................................................................................................................... 6 2.1.2. AAR/ARS Routing Check ......................................................................................................................... 7 2.1.3. Configure Trunk‐to‐Trunk Transfers ....................................................................................................... 7

2.2. ADD NODE NAMES .......................................................................................................................................... 8 2.3. CONFIGURE IP NETWORK REGION ...................................................................................................................... 9 2.4. CONFIGURE SIP SIGNALING GROUP AND TRUNK GROUP ....................................................................................... 10

2.4.1. Add Signaling Group for SIP Trunk ....................................................................................................... 10 2.4.2. Add SIP Trunk Group ............................................................................................................................ 11

2.5. ADMINISTERING NUMBER PLAN ....................................................................................................................... 12 2.5.1. Enable Private Numbering ................................................................................................................... 12 2.5.2. Configure Private Numbering Plan ...................................................................................................... 13

2.6. SAVE TRANSLATIONS ...................................................................................................................................... 14

3. CONFIGURE AVAYA AURA™ SESSION MANAGER .............................................. 15

3.1. ADMINISTER SIP DOMAINS ............................................................................................................................. 16 3.2. DEFINE LOCATIONS ........................................................................................................................................ 18 3.3. ADD CISCO UNIFIED COMMUNICATIONS MANAGER ............................................................................................. 20

3.3.1. Create A Cisco Unified Communications Manager Adaptation ........................................................... 20 3.3.2. Define SIP Entity ................................................................................................................................... 22 3.3.3. Define Entity Link ................................................................................................................................. 23 3.3.4. Define Time Ranges ............................................................................................................................. 24 3.3.5. Define Routing Policy ........................................................................................................................... 25 3.3.6. Define Dial Plan .................................................................................................................................... 26

3.4. ADD AVAYA AURA™ COMMUNICATION MANAGER EVOLUTION SERVER .................................................................... 28 3.4.1. Define SIP Entity ................................................................................................................................... 28 3.4.2. Define Entity Link ................................................................................................................................. 29 3.4.3. Define Routing Policy ........................................................................................................................... 30 3.4.4. Define Dial Plan .................................................................................................................................... 31

4. CONFIGURE CISCO UNIFIED COMMUNICATIONS MANAGER ........................ 32

4.1. LAUNCH CUCM ADMINISTRATION WEB PAGE .................................................................................................... 32 4.2. VERIFY LICENSE ............................................................................................................................................. 32

4.2.1. Phone License Feature ......................................................................................................................... 33 4.2.2. CCM Node License Feature .................................................................................................................. 33

4.3. CONFIGURE CISCO UNIFIED CALLMANAGER ........................................................................................................ 34 4.3.1. Auto‐registration Information ............................................................................................................. 34

4.4. PHONE NETWORK TIME PROTOCOL (NTP) REFERENCE ......................................................................................... 35 4.5. DATE/TIME GROUP INFORMATION ................................................................................................................... 35 4.6. REGION INFORMATION ................................................................................................................................... 36

4.6.1. Verify Default Region is set to use codec G.711 ................................................................................... 36




4.7. SIP TRUNK SECURITY PROFILE .......................................................................................................................... 37 4.8. MEDIA RESOURCES ........................................................................................................................................ 38

4.8.1. Verify Annunciator (ANN) .................................................................................................................... 38 4.8.2. Verify Conference Bridge (CFB) ............................................................................................................ 39 4.8.3. Verify Media Termination Point (MTP) ................................................................................................ 40 4.8.4. Add Music On Hold Audio Source ......................................................................................................... 41 4.8.5. Verify Music On Hold Server (MOH) ..................................................................................................... 42 4.8.6. Define a Media Resource Group (MRG) ............................................................................................... 43 4.8.7. Define a Media Resource Group List (MRGL) ....................................................................................... 44

4.9. CONFIGURE DEFAULT DEVICE POOL .................................................................................................................. 45 4.10. VERIFY STANDARD SIP PROFILE CONFIGURATION ................................................................................................ 46 4.11. ADD SIP TRUNK ............................................................................................................................................ 47 4.12. ADD ROUTE PATTERN ..................................................................................................................................... 49 4.13. ADD PHONES ................................................................................................................................................ 51

4.13.1. Add 7960 SIP Phone ........................................................................................................................ 51 4.13.2. Repeat steps in Section 4.13.1 for second 7960 SIP Phone ............................................................. 59 4.13.3. Add 7960 SCCP Phone ..................................................................................................................... 59

5. VERIFICATION STEPS ................................................................................................... 60

5.1. VERIFY AVAYA AURA™ SESSION MANAGER ......................................................................................................... 60 5.2. VERIFY CISCO UNIFIED COMMUNICATIONS MANAGER .......................................................................................... 61

5.2.1. Enter Cisco Unified CallManager Serviceability ................................................................................... 61 5.2.2. Verify Service Activation ...................................................................................................................... 62 5.2.3. Verify CM Service Are Started .............................................................................................................. 63 5.2.4. Return to the Cisco Unified CallManager Administration .................................................................... 64 5.2.5. Real Time Monitoring Tool ................................................................................................................... 64

5.3. VERIFY AVAYA AURA™ COMMUNICATION MANAGER CONFIGURATION .................................................................... 65 5.4. CALL SCENARIOS VERIFIED ............................................................................................................................... 67

6. ACRONYMS ....................................................................................................................... 68

7. CONCLUSIONS ................................................................................................................. 69

8. REFERENCES .................................................................................................................... 70




1. Introduction These Application Notes describe a sample configuration of a network that uses SIP trunks between Cisco Unified Communication Manager (CUCM) v8.0.3, Avaya Aura™ Session Manager (SM) and an Avaya Aura™ Communication Manager (CM) operating as an Evolution Server. As shown in the Figure 1, the Avaya 96xx Series IP Telephone (H.323) and 2420 Digital Telephone are supported by Avaya Aura™ Communication Manager Evolution Server. The Communication Manager Evolution Server is connected over a SIP trunk to Avaya Aura™ Session Manager, using its virtual SIP network interface. The Cisco Unified Communications Manager supports Cisco 7960 SIP and 7960 SCCP phones and the CUCM is also connected to Avaya Aura™ Session Manager over a SIP trunk. All inter-system calls are carried over this SIP trunk. Avaya Aura™ Session Manager is managed by a separate Avaya Aura™ System Manager. Avaya 9630 IP Telephones configured as SIP endpoints utilize the Avaya Aura™ Session Managers User Registration, which uses Avaya Aura™ Communication Manager Evolution Server to supply feature access to the Avaya SIP phones.

Figure 1: Sample Configuration

For the sample configuration, Avaya Aura™ Session Manager runs on an Avaya S8800 Server, and Avaya Aura™ Communication Manager 6.0 runs on an Avaya S8800 Server with Avaya G650 Media Gateway.




These Application Notes will focus on the configuration of the SIP trunks and call routing. Detailed administration of Communication Manager Evolution Server, SIP endpoints or SIP users will not be described (see the appropriate documentation listed in Section 9). The basic dialing patterns used in the sample configuration used in these Application Notes are as follows: Avaya Core Site Phone(s) (SIP, H.323, Digital) will be able to dial 555-8xxx to reach a

Cisco Phone(s) on the CUCM 8.x cluster or branch site via SIP trunk configured between the Avaya Aura™ Session Manager and the Cisco Unified Communication Manager.

Cisco Phone(s) on the CUCM 8.x cluster or branch site will be able to dial 666-xxxx to reach Avaya Core Site Phone(s) via SIP trunk configured between Cisco Unified Communication Manager and Avaya Aura™ Session Manager.

1.1. Equipment and Software Validated The following equipment and software were used for the sample configuration.

Hardware Component Software Version

S8800 Server with G650 Media Gateway Avaya Aura™ Communications Manager acting as an Evolution Server (R016x.00.0345.0) Patch 1002

S8810 Media Server Session Manager 6.0.0.0.600020 System Manager 6.0 (6.0.0.0.556-3.0.6.1)

Avaya 9630 IP Telephone (SIP) 2.6.0 Avaya 9630 IP Telephone (H.323) S3.002 Avaya 4621SW IP Telephone (H.323) S2.0 Avaya 2420 Digital (DCP) Telephone -- Avaya 6221 Analog Telephone -- Cisco Unified Communications Manager (CallManager)

Product Version: 8.0.3.20000-2 Platform Version : 4.0.0.0-43

Cisco 7960G IP Telephone (SIP) Phone Load: P0S3-8-12-00 Cisco 7060G IP Telephone (SCCP) Phone Load: P00308010200




2. Configure Avaya Aura™ Communication Manager Evolution Server

This section describes the administration of the SIP trunks between the Avaya Aura™ Communication Manager Evolution Server and Avaya Aura™ Session Manager using a System Access Terminal (SAT). These instructions assume the G650 Media Server is already configured on the Avaya Aura™ Communication Manager Evolution Server, as well as ACM H.323, Digital, and analog endpoints. Some administration screens have been abbreviated for clarity.

Verify System Capabilities and Communication Manager Licensing

Administer network region

Administer IP node names

Administer IP interface

Administer SIP trunk group and signaling group

Administer route patterns

Administer numbering plan

After completing these steps, the “save translations” command should be performed.

2.1. Verify System Capabilities and Licensing This section describes the procedures to verify the correct system capabilities and licensing have been configured. If there is insufficient capacity or a required features is not available, contact an authorized Avaya sales representative to make the appropriate changes.

2.1.1. SIP Trunk Capacity Check Issue the “display system-parameters customer-options” command to verify that an adequate number of Registered IP Stations and SIP trunk members are administered for the system as shown below: display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 12000 0 Maximum Concurrently Registered IP Stations: 18000 8 Maximum Administered Remote Office Trunks: 12000 0 Maximum Concurrently Registered Remote Office Stations: 18000 0 Maximum Concurrently Registered IP eCons: 414 0 Max Concur Registered Unauthenticated H.323 Stations: 100 0 Maximum Video Capable Stations: 18000 0 Maximum Video Capable IP Softphones: 18000 3 Maximum Administered SIP Trunks: 24000 85 Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0 Maximum Number of DS1 Boards with Echo Cancellation: 522 0 Maximum TN2501 VAL Boards: 128 2 Maximum Media Gateway VAL Sources: 250 0 Maximum TN2602 Boards with 80 VoIP Channels: 128 0 Maximum TN2602 Boards with 320 VoIP Channels: 128 2 Maximum Number of Expanded Meet-me Conference Ports: 300 0




2.1.2. AAR/ARS Routing Check Verify that ARS and ARS/AAR Dialing without FAC are enabled (on page 3 of system-parameters customer options). display system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES Abbreviated Dialing Enhanced List? y Audible Message Waiting? n Access Security Gateway (ASG)? n Authorization Codes? n Analog Trunk Incoming Call ID? y CAS Branch? n A/D Grp/Sys List Dialing Start at 01? y CAS Main? n Answer Supervision by Call Classifier? n Change COR by FAC? n ARS? y Computer Telephony Adjunct Links? y ARS/AAR Partitioning? y Cvg Of Calls Redirected Off-net? y ARS/AAR Dialing without FAC? y DCS (Basic)? y ASAI Link Core Capabilities? y DCS Call Coverage? y ASAI Link Plus Capabilities? y DCS with Rerouting? y Async. Transfer Mode (ATM) PNC? n Async. Transfer Mode (ATM) Trunking? n Digital Loss Plan Modification? n ATM WAN Spare Processor? n DS1 MSP? y ATMS? n DS1 Echo Cancellation? y Attendant Vectoring? n

2.1.3. Configure Trunk-to-Trunk Transfers Use the “change system-parameters features” command to enable trunk-to-trunk transfers. This feature is needed to be able to transfer an incoming/outgoing call from/to the remote switch back out to the same or another switch For simplicity, the Trunk-to-Trunk Transfer field was set to “all” to enable all trunk-to-trunk transfers on a system wide basis. Note: this feature poses significant security risk, and must be used with caution. change system-parameters features Page 1 of 19 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 10 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? n Music/Tone on Hold: music Type: ext 666-1000 Music (or Silence) on Transferred Trunk Calls? all DID/Tie/ISDN/SIP Intercept Treatment: attd Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred Automatic Circuit Assurance (ACA) Enabled? n Abbreviated Dial Programming by Assigned Lists? n Auto Abbreviated/Delayed Transition Interval (rings): 2 Protocol for Caller ID Analog Terminals: Bellcore




2.2. Add Node Names Using the “change node-names ip” command, add the node-name and IP address for the procr, Avaya Aura™ Session Manager and the Cisco Unified Communications Manager (CUCM), if not already previously added. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address ASM1-SM100 10.80.120.28 ASM2-SM100 10.80.120.30 ATT-MEDPRO-1A11 10.80.111.32 BSM-LSP1 10.80.112.12 BSM1-SM1 10.80.112.15 CUCM8 192.45.130.100 IPOR6 33.1.1.104 MgmtPC1 10.80.51.40 VAL01a08 10.80.111.90 clan-1a03 10.80.111.31 clan-1a04 10.80.111.76 default 0.0.0.0 gateway1 10.80.111.1 procr 10.80.111.73 procr6 :: xfire-1a02 10.80.111.77

Note: To enable procr to be used for signaling, it must be enabled on the system-parameters customer-options form. Using the “change system-parameters customer-options” command, change the Processor Ethernet value to “y” on page 5 of the customer-options form. change system-parameters customer-options Page 5 of 11 OPTIONAL FEATURES Multinational Locations? y Station and Trunk MSP? y Multiple Level Precedence & Preemption? n Station as Virtual Extension? y Multiple Locations? y System Management Data Transfer? n Personal Station Access (PSA)? y Tenant Partitioning? n PNC Duplication? n Terminal Trans. Init. (TTI)? y Port Network Support? y Time of Day Routing? y Posted Messages? n TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Private Networking? y Usage Allocation Enhancements? y Processor and System MSP? y Processor Ethernet? y Wideband Switching? n Wireless? y Remote Office? y Restrict Call Forward Off Net? y Secondary Data Module? n




2.3. Configure IP Network Region Using the “change ip-network-region 1” command, set the Authoritative Domain to the correct SIP domain for the configuration. Verify the Intra-region IP-IP Direct Audio, and Inter-region IP-IP Direct Audio fields are set to “yes”. change ip-network-region 1 Page 1 of 20 IP NETWORK REGION Region: 1 Location: Authoritative Domain: avaya.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 6 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 65535 DIFFSERV/TOS PARAMETERS Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5

Using the “change-ip-network-region 2” command, set the Authoritative Domain to the correct SIP domain for the configuration. Verify the Intra-region IP-IP Direct Audio, and Inter-region IP-IP Direct Audio fields are set to “yes”. change ip-network-region 2 Page 1 of 20 IP NETWORK REGION Region: 2 Location: 1 Authoritative Domain: avaya.com Name: HQ IP Phones MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 6 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 65535 DIFFSERV/TOS PARAMETERS Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5




2.4. Configure SIP Signaling Group and Trunk Group

2.4.1. Add Signaling Group for SIP Trunk Use the “add signaling-group n” command, where “n” is an available signaling group number for the Non-IMS-enabled SIP Trunk to Avaya Aura™ Session Managers In the sample configuration, trunk group “10” and signaling group “10” are used to connect to the Session Manager. Default values can be used for the remaining fields.

Group Type: “sip”

Transport Method: “tcp”

IMS Enabled?: “n”

Near-end Node Name: procr node name from Section 2.2

Far-end Node Name: Session Manager node name from Section 2.2

Near-end Listen Port: “5060”

Far-end Listen Port: “5060”

Far-end Domain: Authoritative Domain from Section 2.3

Session Establishment Timer: “3”

Enable Layer 3 Test: “y”

add signaling-group 10 Page 1 of 1 SIGNALING GROUP Group Number: 10 Group Type: sip IMS Enabled? n Transport Method: tcp Q-SIP? n SIP Enabled LSP? n IP Video? y Priority Video? n Enforce SIPS URI for SRTP? n Peer Detection Enabled? y Peer Server: SM Near-end Node Name: procr Far-end Node Name: ASM1-SM100 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 1 Far-end Domain: avaya.com Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? y Initial IP-IP Direct Media? y H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 9




2.4.2. Add SIP Trunk Group Add the corresponding trunk group controlled by each signaling group using the “add trunk-group n” command, where “n” is an available trunk group number and fill in the indicated fields.

Group Type: “sip”

Group Name: A descriptive name.

TAC: An available trunk access code.

Service Type: “tie”

Signaling Group: The number one of the signaling groups added in Section 2.4.1

Number of Members: The number of SIP trunks to be allocated to route calls to Session Manager Note: The value must be within the limits of the total number of trunks configured in Section 2.1.1.

add trunk-group 10 Page 1 of 21 TRUNK GROUP Group Number: 10 Group Type: sip CDR Reports: y Group Name: to ASM1 COR: 1 TN: 1 TAC: #10 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Member Assignment Method: auto Signaling Group: 10 Number of Members: 25

Once the add command is completed, trunk members will be automatically generated based on the value in the Number of Members field. On page 2, set the Preferred Minimum Session Refresh Interval to 1200. Note: To avoid extra SIP messages, all SIP trunks connected to Session Manager should be configured with a minimum value of 1200. change trunk-group 10 Page 2 of 21 Group Type: sip TRUNK PARAMETERS Unicode Name: auto Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 1200 Delay Call Setup When Accessed Via IGAR? N




On page 3, set Numbering Format to be “private.” Use default values for all other fields. change trunk-group 10 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: private UUI Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Modify Tandem Calling Number: no Show ANSWERED BY on Display? y

2.5. Administering Number Plan SIP Users registered to Session Manager will need to be added to either the private or public numbering table on the Communication Manager Evolution Server. For this sample configuration, private numbering was used.

2.5.1. Enable Private Numbering Use the “change system-parameters customer-options” command to verify that Private Networking is enabled as shown below: display system-parameters customer-options Page 5 of 11 OPTIONAL FEATURES Multinational Locations? y Station and Trunk MSP? y Multiple Level Precedence & Preemption? n Station as Virtual Extension? y Multiple Locations? y System Management Data Transfer? n Personal Station Access (PSA)? y Tenant Partitioning? n PNC Duplication? n Terminal Trans. Init. (TTI)? y Port Network Support? y Time of Day Routing? y Posted Messages? n TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Private Networking? y Usage Allocation Enhancements? y Processor and System MSP? y Processor Ethernet? y Wideband Switching? n Wireless? y Remote Office? y Restrict Call Forward Off Net? y Secondary Data Module? n




2.5.2. Configure Private Numbering Plan Use the “change private-numbering x” command, where “x” is the next available number to create entries in the numbering plan. For this sample configuration, extension numbers starting with 555-XXXX or 666-XXX are used on the Communication Manager Evolution Server. Note: The endpoints configured on the Cisco Unified Communications Manager have 555-8XXX numbers in the sample configuration.

Ext Len: Enter the extension length allowed by the dial plan

Ext Code: Enter leading digit (s) from extension number

Trunk Grp: Enter the SIP Trunk Group number for the SIP trunk between the Evolution Server and Session Manager

Private Prefix: Leave blank unless an enterprise canonical numbering scheme is defined in Session Manager. If so, enter the appropriate prefix.

change private-numbering 1 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len 6 322 10 6 Total Administered: 8 6 333 10 6 Maximum Entries: 540 7 555 10 7 7 662 10 7 7 666 10 7

Verify public-unknown-numbering does not overlap with the private-numbering. In the sample configuration, the public-unknown-numbering plan is not defined as shown below: change public-unknown-numbering 1 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: 0 Maximum Entries: 9999

Note: If an entry applies to a SIP connection to Avaya Aura(tm) Session Manager, the resulting number must be a complete E.164 number.




2.6. Save Translations Configuration of Communication Manager Evolution Server is complete. Use the “save translations” command to save these changes Note: A change on Communication Manager Evolution Server which alters the dial plan will require synchronization between Communication Manager Evolution Server and Session Manager. Also, the SIP phones must be rebooted to pick up the necessary changes. To force synchronization; execute “stop -s sm-mgmt” followed by “start -s sm-mgmt” commands on the Session Manager command line interface.




3. Configure Avaya Aura™ Session Manager This section provides the procedures for configuring the Session Manager and includes the following items:

Administer SIP domain

Define Logical/physical Locations that can be occupied by SIP Entities

For each SIP entity in the sample configuration:

Define SIP Entity

Define Entity Links, which define the SIP trunk parameters used by Avaya Aura™ Session Manager when routing calls to/from SIP Entities

Define Routing Policies, which control call routing between the SIP Entities

Define Dial Patterns, which govern to which SIP Entity a call is routed

Define the Communication Manager Evolution Server as an administration entity

Configuration is accomplished by accessing the browser-based GUI of Avaya Aura™ System Manager, using the URL “http://<ip-address>/SMGR”, where “<ip-address>” is the IP address of Avaya Aura™ System Manager. Log in with the appropriate credentials and accept the Copyright Notice.

Expand Routing on the left side of Navigation Menu. Select a specific item such as SIP Domains. When the specific item is selected, the color of the item will change to blue as shown below:




3.1. Administer SIP Domains Add the SIP domains for the communications infrastructure. Select Routing → Domains on the left menu and click the New button under the Domain Management section.

Fill in the following:

Name: The authoritative domain name

Notes: Descriptive text (optional)

Click Commit to save. The screen below shows the information for Communication Manager Evolution Server in the sample configuration.




The screen below shows the information for sample configuration which includes SIP domains for “avaya.com” and “cucm.com”.




3.2. Define Locations Locations can be used to identify logical and/or physical locations where SIP Entities reside for purposes of bandwidth management and call admission control. To add a location, select Routing → Locations on the left menu and click on the New button (not shown) on the right. Under General, enter:

Name: A descriptive name.

Notes: Add a brief description. The remaining fields under General can be filled in to specify bandwidth management parameters between Session Manager and this location. These were not used in the sample configuration, and reflect default values. Note also that although not implemented in the sample configuration, routing policies can be defined based on location. Under Location Pattern:

IP Address Pattern: An IP address pattern used to identify the location

Notes: Add a brief description

Click Commit to save. The screen below shows the information for Communication Manager Evolution Server which is in the 10.80.111.xxx subnet.




Add locations for all of the following:

Avaya Aura™ Communication Manager Evolution Server (Location 1 Subnet 10.80.111.x)

Avaya Aura™ Session Manager & Avaya Aura™ System Manager (Location 1 Subnet 10.80.120.x)

Cisco Unified Communications Manager (CUCM Location, 192.45.130.*)




3.3. Add Cisco Unified Communications Manager The following section captures relevant screens for configuring Cisco Unified Communications Manager in Session Manager applicable for the sample configuration.

3.3.1. Create A Cisco Unified Communications Manager Adaptation Select Routing → Adaptations on the left menu and click on the New button (not shown) on the right. Configure the following adaptation settings.

Under General:|

Adaptation Name: Add an identifier for the Cisco Unified Communications Manager. CUCM 8.x was used in the sample configuration below.

Module Name: Select “CiscoAdapter” from the drop down list. If CiscoAdapter is not in the list, select <click to add module> and type in the name “CiscoAdapter.”

Module Parameter: Enter iosrcd=avaya.com odstd=192.45.130.100 Note: The iosrcd parameter replaces the domain in the P-Asserted-Identity header and calling part of the History-Info header with the given value for ingress only. The odstd parameter replaces the domain in the Request-URI and Notify/message-summary body with the given value for egress only.

Egress URI Parameters: Leave blank.

Notes: Enter a brief description. (optional)




Under Digit Conversion for Incoming Call to SM:

Matching Pattern: Enter a numberic number patter to match on.

Min: Enter the minimum number of digits for the dialed number.

Max: Enter the maximum number of digits for the dialed number.

Delete Digits: 0

Address to Modify: both

Under Digit Conversion for Outgoing Call from SM:

Matching Pattern: Enter a numberic number patter to match on.

Min: Enter the minimum number of digits for the dialed number.

Max: Enter the maximum number of digits for the dialed number.

Delete Digits: 3

Address to Modify: both

Click Commit to save.




3.3.2. Define SIP Entity Select Routing → SIP Entities on the left menu and click on the New button (not shown) on the right.

Under General: Name: The Cisco Unified Communications Manager Name. In

this sample configuration we used CUCM 8.x, since we are using a 8.0.3 version of Cisco’s Unified Communications Manager

FQDN or IP Address: The IP Address of the CUCM

Type: “Other”

Adaptation: Select the newly created adaptation “CUCM 8.x”, added in Section 3.3.1

Location: Select from the drop-down list the Location added in Section 3.2 Note: since location-based routing was not used in the sample configuration, selecting a value for location field is optional.

Click Commit to save. The following screen shows addition of Cisco Unified Communications Manager using its configured IP address.




3.3.3. Define Entity Link Select Routing → Entity Links on the left menu and click on the New button (not shown) on the right.

Name: Enter an identifier for the Cisco Unified Communication Manager. CUCM 8.x was used in the sample configuration below.

SIP Entity 1: From the drop-down, select the appropriate Session Manager. (SM1)

Protocol: Select the protocol to use. TCP was used in the sample configuration.

Port: From the drop-down, select the correct port for the Session Manager. (5060 was used in the sample configuration)

SIP Entity 2: From the drop-down, select the SIP Entity added for the Cisco Unified Communications Manager (CUCM 8.x).

Port: From the drop-down, select the correct port for the Communication Manager. (5060 was used in the sample configuration)

Trusted Checkbox: Check the trusted checkbox to accept the SIP Entity as a trusted host.

Notes: Add a brief description.

Click Commit to save. The following screen shows the entity link defined for the CUCM to the Session Manager.




3.3.4. Define Time Ranges Select Routing → Time Ranges on the left menu and click on the New button (not shown) on the right.

Name: Add an identifier to define the time range. 24/7 was used in the sample configuration.

Days of the Week: Check all boxes under the days of the week.

Start Time: Enter “00:00”

Stop Time: Enter “23:59”

Notes: A brief description of the time range. “Time Range 24/7” was used in the sample configuration.





3.3.5. Define Routing Policy Select Routing → Routing Policies on the left menu and click on the New button (not shown) on the right.

Under General: Name: Add an identifier to define the routing policy for the

Cisco Unified Communications Manager.

Notes: Add a brief description

Under SIP Entity as Destination: Click on the Select button and the SIP Entity List page opens.

Select the entry of the CUCM 8.x that was added in Section 3.3.2 and click on Select.

Verify the selected SIP Entity displays on the Routing Policy Details page.

Click on Commit to save.




3.3.6. Define Dial Plan Select Routing → Dial Patterns on the left menu and click on the New button (not shown) on the right.

Under General: Pattern: Add dial pattern to match on for numbers located on

CUCM 8.x. For the sample configuration below, 5558xxx was used.

Min: Enter the minimum number digits that must to be dialed. Sample Configuration: (7 digits)

Max: Enter the maximum number digits that may be dialed. Sample Configuration: (7 digits)

SIP Domain: From the drop-down, select the SIP Domain added in Section 3.1 or select “All” if the system can accept incoming calls from all SIP domains. In the sample configuration “ALL” was selected for the SIP Domain.

Notes: Add a brief description. (optional)




Under Originating Locations and Routing Policies:

Click on Add

Originating Location Name: Select the desired location. In the sample configuration, Apply The Selected Routing Policies to All Origination Locations was selected.

Routing Policies: Select the routing policy defined for Cisco Unified

Communications Manager (CUCM 8.x).





3.4. Add Avaya Aura™ Communication Manager Evolution Server The following section captures relevant screens for configuring Avaya Aura™ Communication Manager Evolution Server in Session Manager applicable for the sample configuration.

3.4.1. Define SIP Entity Select Routing → SIP Entities on the left menu and click on the New button (not shown) on the right.

Under General:

Name: Add an identifier for the Avaya Aura™ Communication Manager Evolution Server. The sample configuration used “S8800-CM 6.0 ES”.

FQDN or IP Address: Enter the IP Address or Fully Qualified Domain Name of the Communication Manager Evolution Server (procr).

Type: Select CM.


Location: Select the location of the Evolution Server as defined in section 3.2, “Location 1 Subnet 10.80.111.x”.

Time Zone: Select the correct time zone. The sample configuration used “America/Denver.”





3.4.2. Define Entity Link Select Routing → Entity Links on the left menu and click on the New button (not shown) on the right.

Name: Enter an identifier for the Communication Manager Evolution Server.

SIP Entity 1: From the drop-down, select the appropriate Session Manager, SM1.

Protocol: From the drop-down, select the required protocol, TCP.

Port: From the drop-down, select the correct port to use for the Session Manager, 5060.

SIP Entity 2: From the drop-down, select one of the SIP Entities added in Section 3.4.1 for the Communication Manager Evolution Server.

Port: From the drop-down, select the correct port to use for the Communication Manager Evolution Server, 5060.

Trusted: Check box to allow as a trusted host.



The following screen shows the entity link defined for the Communication Manager Evolution Server.




3.4.3. Define Routing Policy Select Routing → Routing Policies on the left menu and click on the New button (not shown) on the right. Under General:

Name: Add an identifier to define the routing policy for the Communication Manager Evolution Server.

Disabled: Leave unchecked.


Under SIP Entity as Destination:

Click on Select button and the “SIP Entity List” page opens.

Select one of the SIP Entities added in Section 3.4.1 for the Communication Manager Evolution Server.

The selected SIP Entity displays on the Routing Policy Details page.


Shown below is the updated screen for the sample configuration.




3.4.4. Define Dial Plan Select Routing → Dial Patterns on the left menu and click on the New button (not shown) on the right.

Under General:

Pattern: Add the dial patterns associated with extensions on the Communication Manager Evolution Server.

Min: Enter the minimum number digits that must to be dialed.

Max: Enter the maximum number digits that may be dialed.

SIP Domain: From the drop-down, select one of the SIP Entities added in Section 3.4.1.


Under Origination Locations and Routing Policies:

Click on Add and the “Locations and Routing Policy List” page opens.

Locations: Select the desired location. Note: since location-based routing was not used in the sample configuration, selecting a value for location field is optional.

Routing Policies: Select the defined in Section 3.4.3 for Communication Manager Evolution Server.

. Click on Commit to save.

Shown below is the updated screen for the sample configuration.




4. Configure Cisco Unified Communications Manager This section describes the relevant configuration of the Cisco Unified Communications Manager used to verify these Application Notes. Please consult the product documentation referenced in Section 9 for additional information.

4.1. Launch CUCM Administration Web Page The Cisco Unified Communications Manager is configured using its web based administration GUI. Enter the following URL address format in Internet Explorer (IE) browser to access the administration page: https://<IP Address of CUCM>:8443/ccmadmin/showHome.do Log in using the appropriate Username and Password.

4.2. Verify License From the administration tool bar, select SystemLicensingLicense Unit Report




4.2.1. Phone License Feature Under Phone License Feature:

Check that the License Server has Units Authorized.

Note: These are often referred to in Cisco documentation as Device License Units (DLUs).

Check that the numbers of Units Remaining are adequate to support the phone devices to be added to the system. Note: Cisco phone devices require different amounts of DLUs per device based on type and model of device being used. Example: 7960 4 - DLUs 7970 5 – DLUs Third Party (Basic) 3 – DLUs Third Party (Adv.) 6 – DLUs IP Communicator 3 – DLUs For the sample configuration described in this document the following are used:

2 – Cisco 7960 (SIP) phones 1 – Cisco 7960 (SCCP)

Therefore, the sample configuration will require 12 available DLUs; [3 (7960 Phones) x 4 (DLUs/device) = 12 DLUs required].

4.2.2. CCM Node License Feature Under CCM Node License Feature:

Check the License Server has Units Authorized.

Check the Units Used and Remaining are correct for the number of node in the CallManager cluster. Note: In the sample configuration described in this document only one node (publisher) will be used. Therefore, only one unit is used with zero remaining.




4.3. Configure Cisco Unified CallManager Select System → Cisco Unified CM. Click on Find and select the CallManager (Publisher).

4.3.1. Auto-registration Information Under Auto-registration Information:

Check the check-box for “Auto-registration Disabled on this Cisco Unified CallManager.” Note: This will allow phones and Directory Number (DN) lines to be added manually for this sample configuration.

Under Cisco Unified CallManager TCP Port Settings for this Server:

SIP Phone Port: 5060

SIP Phone Secure Port: 5061

Click on Save.




4.4. Phone Network Time Protocol (NTP) Reference Select System → Phone NTP Reference. Click on Find and select Add New.

IP Address: Enter the IP Address of the NTP source.

Description: Enter a brief description of the NTP source.

Mode: Select Directed Broadcast

Click on Save.

4.5. Date/Time Group Information The date and time configuration controls the Time Zone, Display Format and the Network Time Protocol (NTP) References used by registered devices. Having all devices synced with the same NTP source makes debugging much easier to reference between systems and phone traces. This configuration step is optional, but recommended if troubleshooting is needed. Select System → Date/Time Group.

Group Name: Select the default Date/Time Group, CMLocal

Time Zone: Select the appropriate Time Zone

Date Format: Select the appropriate Date Format

Time Format: Select the appropriate Time Format




Click on Add Phone NTP References button. A popup window will appear with a list of configured NTP references.

Select the NTP reference by checking the box next to the desired NTP and clicking on Add Selected button.

Click on Save.

4.6. Region Information Select System → Region. Click on Find and select the Default region.

4.6.1. Verify Default Region is set to use codec G.711 Verify Default region has audio codec G.711 and Video Call Bandwidth 384

configured.

If the region codec or video bandwidth settings need to be changed, select Default under the “Modify Relationship to other Regions” and then select G.711 from the drop down list under “Audio Codec” and select the radio button for the desired “Video Call Bandwidth.”




Note: The default region codec set was changed to G.729 during testing focused on the G.729 codec set and then changed back to G.711 when finished.

4.7. SIP Trunk Security Profile Select System → Security → SIP Trunk Security Profile

Click on Find to list the available profiles

Add a new profile by clicking on the Add New button

Configure the new profile with settings listed below:




4.8. Media Resources In order to support some of the supplementary service features like Transfer, Conferencing, Music on Hold (MOH), call Annunciation, etc., media resource groups and lists need to be configured using default software Annunciators (ANN), Conference Bridges (CFBs), Music on Hold (MOH) and Media Termination Points (MTPs). This section will verify the default installed ANN, CFB, MOH and MTP, which will be used to create a new Media Resource Group (MRG), and then a Media Resource Group List (MRGL), which will be used in additional configurations of the SIP Trunk and Phone Devices.

4.8.1. Verify Annunciator (ANN) Select Media Resources → Annunciator

Click on Find to list the available annunciators.

A default annunciator should be available and registered with the CUCM publisher listing its assigned IP address.

Select the default ANN. (In the sample configuration below the annunciator is named ANN_2; this name may be different on other systems).

Verify the Device Pool assigned is “Default.”




4.8.2. Verify Conference Bridge (CFB) Select Media Resources → Conference Bridge

Click on Find to list the available conference bridges.

A default conference bridge should be available and registered to the CUCM publisher listing its assigned IP address.

Select the default CFB. (In the sample configuration below the conference bridge is named CFB_2; this name may be different on other systems).





4.8.3. Verify Media Termination Point (MTP) Select Media Resources → Media Termination Point

Click on Find to list the available MTPs.

A default MTP should be available and registered to the CUCM publisher listing its assigned IP address.

Select the default MTP. (In the sample configuration below the MTP is named MTP_2; this name may be different on other systems).





4.8.4. Add Music On Hold Audio Source Select Media Resources → Music On Hold Audio Sources

Click on Find to list the available MOH stream numbers.

Verify a MOH is listed and configured to use the SampleAudioSource; if there is no MOH defined click on Add New button to create one.

Verify the following in the MOH configuration page:

MOH Audio Source File: SampleAudioSource

MOH Audio Source Name: SampleAudioSource

Play continuously is checked

Allow multicasting is checked

Click on Save.




4.8.5. Verify Music On Hold Server (MOH) Select Media Resources → Music On Hold Server

Click on Find to list the available MOH Servers.

A default MOH server should be available and registered to the CUCM publisher listing its assigned IP address.

Select the default MOH server. (In the sample configuration below the MOH server is named MOH_2; this name may be different on other systems).





4.8.6. Define a Media Resource Group (MRG) Select Media Resources → Media Resource Group Select Add New.

Name: Enter “MRG_1”

Description: Enter a brief description.

Under Devices for this Group:

Select the following devices from the “Available Media Resources” window and click on the down arrow () move the selected resource to the “Selected Media Resources” window:

ANN_2

CFB_2

MOH_2

MTP_2

Note: The resource names in the Available Media Resource window may be different that those listed in the sample configuration below. Please use the default names of the resources verified in the media verifications in previous steps.

Click on Save.




4.8.7. Define a Media Resource Group List (MRGL) Select Media Resources → Media Resource Group List Select Add New.

Name: Enter “MRGL_1”

Under Media Resource Groups for this List:

In the “Available Media Resource Groups” window, select MRG_1 and click on the down arrow () to move the selected MRG to the lower window, “Selected Media Resource Groups.”

Click on Save.




4.9. Configure Default Device Pool Select System → Device Pool Click on Find to list the device pools and verify the “Default” device pool settings:

Device Pool Name: Default

Date/Time Group: CMLocal

Media Resource Group List: MRGL_1

Click on Save.




4.10. Verify Standard SIP Profile Configuration Select Device → Device Settings Select SIP Profile.

Select Find to list configured SIP Profiles and select Standard SIP Profile. Verify the following settings:

Name: Standard SIP Profile

Default MTP Telephony Event Payload Type: Enter 101

Disable Early Media on 180: Box Unchecked

Conference Join Enabled: Box Checked

RFC 2543 Hold: Box Unchecked

Semi Attended Transfer: Box Checked

Fall back to local RSVP: Box Checked




4.11. Add SIP Trunk Select Device → Trunk Click on Add New.

Trunk Type: Select “SIP Trunk”

Device Protocol: Select “SIP”

Trunk Service Type: None (Default)

Click the Next button Under Device Information:

Device Name: Enter “SIP-Trunk-To-SM6.0”

Description: Enter a brief description

Device Pool: Select “Default”

Media Resource Group List: Select “MRGL_1”

Media Termination Point Required: Uncheck box

Note: It was noted during setup and verification that if the “Media Termination Point Required” was checked; shuffling on the Avaya components were restricted to using IP-TDM for all connections whereas with this item unchecked, shuffling worked correctly for all calls.




Under SIP Information:

Destination Address: Enter the IP address of the Avaya Aura™ Session Manager

Destination Port: 5060

Presence Group: Select “Standard Presence Group”

SIP Trunk Security Profile: Select “Avaya SIP Trunk Profile”

SIP Profile: Select “Standard SIP Profile”

DTMF Signaling Method: Select RFC 2833

Click on Save.




4.12. Add Route Pattern Select Call Routing → Route/Hunt → Route Pattern Click on Find to list configured route patterns. Click on Add New. Under Pattern Definition:

Route Pattern: Enter “666XXXX”

Route Partition: Select <None>

Description: Enter a brief description for the route pattern

MLPP Precedence: Select “Default”

Gateway/Route List: Select “SIP-Trunk-To-SM6.0”; the SIP trunk configured in Section 4.11.

Route Option: Select the radio button , “Route this pattern”

Call Classification: Select “OffNet”

Provide Outside Dial Tone: Check the box




Under Calling Party Transformations:

Use Calling Party’s External Phone Number Mask: Check this box

Calling Party Transform Mask: Enter “555XXXX”

Calling Line ID Presentation: Select “Allowed”

Calling Name Presentation: Select “Allowed”

Click on Save.




4.13. Add Phones

4.13.1. Add 7960 SIP Phone Select Device → Phone Click on Find to list all configured phones. Click on Add New.

Phone Type Select “Cisco 7960”

Click on Next

Device Protocol: Select “SIP”

Click on Next




Under Device Information:

MAC Address: Enter MAC address of the phone. Note: The MAC address can be found on the center label located on the lower back side of the phone.

Description: Enter a brief description of the phone. In the sample configuration below we have selected to use “CUCM 8.x Phone A1 – SIP” to describe this phone.

Device Pool: Select “Default”

Phone Button Template: Select “Standard 7960 SIP”

Common Phone Profile: Select “Standard Common Phone Profile”

Media Resource Group List: Select “MRGL_1”

User Hold MOH Audio Source: Select “SampleAudioSource”

Network Hold MOH Audio Source: Select “SampleAudioSource”

Location: Select “Hub_None”







Under Protocol Specific Information:

Device Security Profile: Select “Cisco 7960 – Standard SIP Non-Secure Profile”

SIP Profile: Select “Standard SIP Profile”

Click on Save.




4.13.1.1 Add Line 1 DN

From the Phone Configuration page, select “Line [1] – Add a new DN” under the Association Information section on the left. The Directory Number Configuration Page will open.

From the Directory Number Configuration page enter the following information:

Under Directory Number Information:

Directory Number: Enter the directory number to use for this line on the phone. In the sample configuration 8001 is used.

Description: Enter a brief description for the DN.

Alerting Name: Enter what should be displayed on the alerting phones display when a call is being placed to that phone. Note: The ASCII Alerting Name field will be auto populated after entering text in the Alerting Name field and pressing enter.




Under Directory Number Settings:

User Hold MOH Audio Source: Select “SampleAudioSource”

Network Hold MOH Audio Source: Select “SampleAudioSource”

Under Line 1 on Device SEP<MAC Address>:

Display (Internal Caller ID): Enter “Phone A1-1.”

Note: The ASCII Display (Internal Caller ID) will be auto populated.

Line Text Label: Enter “Phone A1-1 8001.” Note: This will be the text that appears on the LCD of the phone next to the Line 1 extension. Also, the ASCII Line Text Label will be auto populated.

External Phone Number Mask: Enter 555XXXX




Under Forwarded Call Information Display on Device SEP<MAC Address>:

Verify the Forward Call Information Display has the following items checked:

Caller Name

Caller Number

Redirected Number

Dialed Number

Click on Save.




4.13.1.2 Add Line 2 DN

Add a second DN with number 8011 to Line 2 of the phone configured in Section 4.13.1.1. Follow the same steps in Section 4.13.1.1 for the Line 2 DN configuration using the appropriate values. The sample configuration for Line 2 DN is shown below:




4.13.2. Repeat steps in Section 4.13.1 for second 7960 SIP Phone Phone Name: Phone A2 – SIP

Line 1 DN: 8002 Line 2 DN: 8012

4.13.3. Add 7960 SCCP Phone Add the 7960 SCCP phone using the same steps described in Section 4.13.1 using the phone type of SCCP. The remaining configuration steps for both the phone and DNs will remain the same described Section 4.13.1 using SCCP in place of SIP for entered text fields. The Phone and DN configuration are as follows: Phone Name: Phone A3 – SCCP

Line 1 DN: 8003 Line 2 DN: 8013




5. Verification Steps

5.1. Verify Avaya Aura™ Session Manager In the Avaya Aura™ System Manager 6.0 administration GUI, select Elements → Session Manager → System Status → SIP Entity Monitoring from the left menu panel.

Verify as shown below that none of the links for SIP entities are down, indicating that they are all reachable for routing.




Select the SIP Entity Name for the Cisco Unified Communications Manager 8.x (CUCM 8.x) and verify the connection status is “Up”, as shown below:

5.2. Verify Cisco Unified Communications Manager

5.2.1. Enter Cisco Unified CallManager Serviceability From the Navigation drop down in the upper right hand corner of the CUCM Administration screen select Cisco Unified CallManager Serviceability




5.2.2. Verify Service Activation Select Tools → Service Activation Select Server name for publisher. Check the checkbox next to “Check All Services”

Click Save

An information window noting that “Activating/Deactivating services will take a while… Please wait for the page to refresh.” Click OK to accept.

Verify all services show “Activated.”




5.2.3. Verify CM Service Are Started Select Tools → Control Center – Feature Services Select Server name for the publisher. Verify the following feature services have been started, if not select the radio button next to

the service and click the green arrow at the top to start the service.

Cisco CallManager

Cisco Tftp

Cisco IP Voice Media Streaming App

Cisco Extended Functions




5.2.4. Return to the Cisco Unified CallManager Administration In the Navigation drop down box in the upper right, select the Cisco Unified CallManager Administration to return to the main administration web page.

5.2.5. Real Time Monitoring Tool The Real Time Monitoring Tool (RTMT) can be used to monitor events on Cisco Unified Communications Manager. This tool can be downloaded by selecting Application Plugins from the top menu of the Cisco Unified CM Administration Web interface. For further information on this tool, please consult Reference Error! Reference source not found.. The following screen shows where user can view and perform real time data capture.




5.3. Verify Avaya Aura™ Communication Manager Configuration Verify the status of the SIP trunk group by using the “status trunk n” command, where “n” is the trunk group number. Verify that all trunks are in the “in-service/idle” state as shown below: status trunk 10 Page 1 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy 0010/001 T00001 in-service/idle no 0010/002 T00002 in-service/idle no 0010/003 T00003 in-service/idle no 0010/004 T00004 in-service/idle no 0010/005 T00005 in-service/idle no 0010/006 T00006 in-service/idle no 0010/007 T00007 in-service/idle no 0010/008 T00008 in-service/idle no 0010/009 T00009 in-service/idle no 0010/010 T00010 in-service/idle no 0010/011 T00011 in-service/idle no 0010/012 T00012 in-service/idle no 0010/013 T00013 in-service/idle no 0010/014 T00014 in-service/idle no

Verify the status of the SIP signaling groups by using the “status signaling-group n” command, where “n” is the signaling group number. Verify the signaling group is “in-service” as indicated in the “Group State” field shown below: status signaling-group 10 STATUS SIGNALING GROUP Group ID: 10 Group Type: sip Group State: in-service




Use the CM SAT command, “list trace tac #”, where “tac #”, e.g.#10 as shown in Section 2.4.2, is the trunk access code allowing it to trace trunk group activity for the SIP trunk between the Session Manager and the Communication Manager Evolution Server as shown below:

list trace tac #10 Page 1 LIST TRACE time data 20:35:39 TRACE STARTED 10/03/2010 CM Release String cold-00.0.345.0-18444 20:35:52 SIP<INVITE sip:[email protected]:5060 SIP/2.0 20:35:52 active trunk-group 10 member 1 cid 0x237 20:35:52 dial 6664006 20:35:52 term station 6664006 cid 0x237 20:35:52 SIP>SIP/2.0 180 Ringing 20:35:54 SIP>SIP/2.0 200 OK 20:35:54 active station 6664006 cid 0x237 20:35:54 SIP<ACK sip:[email protected]:5060;transport=tcp SIP 20:35:54 SIP</2.0 20:35:54 G711MU ss:off ps:20 rgn:1 [10.80.60.164]:28554 rgn:1 [10.80.111.77]:28276 20:35:54 xoip options: fax:Relay modem:off tty:US uid:0x50001 xoip ip: [10.80.111.77]:28276 press CANCEL to quit -- press NEXT PAGE to continue

Use the CM SAT command, “list trace station xxx”, where “xxx” is the extension number of the Avaya 9630 SIP telephone as shown below:

list trace station 6664006 Page 1 LIST TRACE time data 20:37:23 TRACE STARTED 10/03/2010 CM Release String cold-00.0.345.0-18444 20:37:30 SIP<INVITE sip:[email protected]:5060 SIP/2.0 20:37:30 Calling party trunk-group 10 member 1 cid 0x239 20:37:30 Calling Number & Name 5558001 Phone A1-1 20:37:30 dial 6664006 20:37:30 term station 6664006 cid 0x239 20:37:30 dial 866640 route:AAR 20:37:30 term trunk-group 10 cid 0x23a 20:37:30 dial 86664006 route:AAR 20:37:30 route-pattern 10 preference 1 cid 0x23a 20:37:30 seize trunk-group 10 member 22 cid 0x23a 20:37:30 Calling Number & Name NO-CPNumber NO-CPName 20:37:30 Proceed trunk-group 10 member 22 cid 0x23a 20:37:30 SIP>SIP/2.0 180 Ringing 20:37:30 Alert trunk-group 10 member 22 cid 0x23a




5.4. Call Scenarios Verified Verification scenarios for the configuration described in these Application Notes included the following call scenarios: Basic Calls to/from Avaya Analog, Digital, H.323 and SIP endpoints on the Avaya Aura™

system from/to Cisco SCCP and SIP endpoints via a SIP trunk connection between Avaya Aura™ Session Manager and Cisco Unified Communications Manager using the G.711 codec set. These calls were verified using both shuffled and non-shuffled modes.

Basic Calls described above using the G.729 codec set. These calls were verified using both shuffled and non-shuffled modes.

Basic Calls described above using different codec sets on either side verifying the ability to perform codec negotiation between the two systems. These calls were verified using both shuffled and non-shuffled modes.

Call abandonment was verified between the two systems.

Supplementary Service Calling Features were verified between the two systems using Avaya’s H.323 and SIP endpoints with Cisco’s SCCP and SIP endpoints via a SIP trunk connection. The following supplementary services were verified:

Hold Consultative Hold – Soft Hold Hard Hold Unattended Transfer Attended Transfer Call Forwarding – Call Forward All (CFA) Conference Calling Number Block

Display Name and Number were verified on applicable phones during verification of basic

and advance calling with supplementary service calling features.




6. Acronyms

AAR Automatic Alternative Routing (Routing on Communication Manager)

ARS Alternative Routing Service (Routing on Communication Manager)

CLAN Control LAN (Control Card in Communication Manager) CUCM Cisco Unified Communications Manager (CallManager) DCP Digital Communications Protocol DLU Cisco Device License Unit(s) DN Directory Number DNS Domain Naming Resolution DTMF Dual Tone Multi Frequency FQDN Fully Qualified Domain Name (hostname for Domain Naming

Resolution) GUI Graphical User Interface IP Internet Protocol LAN Local Area Network NTP Network Time Protocol RTP Real Time Protocol SAT System Access Terminal (Communication Administration

Interface) SIL Solution Interoperability Lab SIP Session Initiation Protocol SM Avaya Aura™ Session Manager SMGR System Manager (used to configure Session Manager) SSH Secure Shell SSL Secure Socket Layer TAC Trunk Access Code (Communication Manager Trunk Access) TCP Transmission Control Protocol TCP/IP Transmission Control Protocol/Internet Protocol TLS Transport Layer Security URE User Relation Element URL Uniform Resource Locator WAN Wide Area Network




7. Conclusions As illustrated in these Application Notes, Avaya Aura™ Session Manager, and Avaya Aura™ Communication Manager acting as an Evolution Server can interoperate with Cisco Unified Communications Manager using SIP trunks via Avaya Aura™ Session Manager. These Application Notes describe how to configure a network that uses a SIP trunk between Cisco Unified Communications Manager 8.0.3 and Avaya Aura™ Session Manager 6.0. Interoperability tests included making successful bi-directional calls between several different types of endpoints, using various supplementary call processing features including hold, transfer and conference interworking across a SIP trunk connection. The following is a list of interoperability items to note: Calls fail to shuffle on the Avaya Aura™ Communications Manager Evolution Server and

Avaya Aura™ Session Manager when the Cisco SCCP phone was the originator of basic calls across the interconnecting SIP trunk to either an Avaya H.323 or SIP endpoint. The calls were successful, but the status showed IP-TDM was being used on the Communication Manager Evolution Server.

Call failed to shuffle when the calling phone was an Avaya H.323 to the called phone, Cisco SCCP phone. The call was successful, but the status showed IP-TDM was being used on the Communication Manager Evolution Server.

In order to support conferencing on the Cisco Unified Communications Manager using Cisco SIP and SCCP phones, the default media resources needed to be configured into a media resource group and media resource group list as show in Section 4.8. This media resource group list was then used in configurations of the SIP trunk and the IP Phones.

In the specific call scenario, when the Cisco 7960 SIP phone calls an Avaya H.323 phone, then the call is placed on hold from the Cisco phone, and when the Cisco phone resumes the call, one-way talk path was experienced. Audio could be heard on the Cisco phone from the Avaya phone, but the Avaya phone had no audio path from the Cisco phone. All other Hold/Resume scenario’s worked correctly between Avaya and Cisco phones.

In the specific call scenario, when the Cisco 7960 SIP phone calls an Avaya H.323 phone and conferences a Cisco 7960 SCCP phone, there is one-way audio between the Avaya H.323 phone and the two Cisco phones (SIP, SCCP). The Avaya H.323 phone cannot hear any audio from the conference, but the Cisco phones can hear audio from the Avaya phone as well as audio between the Cisco phones. All other conference scenario’s worked correctly.




8. References This section references the product documentation relevant to these Application Notes.

Avaya Aura™ Session Manager

[1] Avaya Aura™ Session Manager Overview, Doc ID 03-603323, Issue 3, Release 6.0, June 2010, available at http://support.avaya.com.

[2] Installing and Configuring Avaya Aura™ Session Manager, Doc ID 03-603473, Issue 1.0, June 2010, available at http://support.avaya.com.

[3] Installing and Upgrading Avaya Aura™ System Manager, Release 6.0, June 2010, available at http://support.avaya.com.

[4] Maintaining and Troubleshooting Avaya Aura™ Session Manager, Doc ID 03-603325, Issue 1.0, Release 6.0, June 2010, available at http://support.avaya.com.

Avaya Aura™ Communication Manager 6.0

[5] SIP Support in Avaya Aura™ Communication Manager Running on Avaya S8xxx Servers, Doc ID 555-245-206, May 2009, available at http://support.avaya.com.

[6] Administering Avaya Aura™ Communication Manager, Doc ID 03-300509, Issue 6.0, Release 6.0, June 2010, available at http://support.avaya.com.

[7] Administering Avaya Aura™ Communication Manager Server Options, Doc ID 03-603479, Issue 2, Release 6.0, June 2010, available at http://support.avaya.com

Cisco Unified Communications Manager 8.0

[8] Cisco Unified Communications Manager Documentation Guide for Release 8.0(3), Doc ID 78-19553-01, July 7, 2010, available at http://www.cisco.com.

Avaya Application Notes

[9] Configuring 9600 Series SIP Phones on Avaya Aura™ Session Manager Release 5.2, available at http://www.avaya.com.




©2010 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 [email protected]

Documents

Configuring SIP Trunks among Cisco Unified