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

Transcription

1 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. 1 of 71

2 Table of Contents 1. INTRODUCTION EQUIPMENT AND SOFTWARE VALIDATED CONFIGURE AVAYA AURA COMMUNICATION MANAGER EVOLUTION SERVER VERIFY SYSTEM CAPABILITIES AND LICENSING SIP Trunk Capacity Check AAR/ARS Routing Check Configure Trunk to Trunk Transfers ADD NODE NAMES CONFIGURE IP NETWORK REGION CONFIGURE SIP SIGNALING GROUP AND TRUNK GROUP Add Signaling Group for SIP Trunk Add SIP Trunk Group ADMINISTERING NUMBER PLAN Enable Private Numbering Configure Private Numbering Plan SAVE TRANSLATIONS CONFIGURE AVAYA AURA SESSION MANAGER ADMINISTER SIP DOMAINS DEFINE LOCATIONS ADD CISCO UNIFIED COMMUNICATIONS MANAGER Create A Cisco Unified Communications Manager Adaptation Define SIP Entity Define Entity Link Define Time Ranges Define Routing Policy Define Dial Plan ADD AVAYA AURA COMMUNICATION MANAGER EVOLUTION SERVER Define SIP Entity Define Entity Link Define Routing Policy Define Dial Plan CONFIGURE CISCO UNIFIED COMMUNICATIONS MANAGER LAUNCH CUCM ADMINISTRATION WEB PAGE VERIFY LICENSE Phone License Feature CCM Node License Feature CONFIGURE CISCO UNIFIED CALLMANAGER Auto registration Information PHONE NETWORK TIME PROTOCOL (NTP) REFERENCE DATE/TIME GROUP INFORMATION REGION INFORMATION Verify Default Region is set to use codec G of 71

3 4.7. SIP TRUNK SECURITY PROFILE MEDIA RESOURCES Verify Annunciator (ANN) Verify Conference Bridge (CFB) Verify Media Termination Point (MTP) Add Music On Hold Audio Source Verify Music On Hold Server (MOH) Define a Media Resource Group (MRG) Define a Media Resource Group List (MRGL) CONFIGURE DEFAULT DEVICE POOL VERIFY STANDARD SIP PROFILE CONFIGURATION ADD SIP TRUNK ADD ROUTE PATTERN ADD PHONES Add 7960 SIP Phone Repeat steps in Section for second 7960 SIP Phone Add 7960 SCCP Phone VERIFICATION STEPS VERIFY AVAYA AURA SESSION MANAGER VERIFY CISCO UNIFIED COMMUNICATIONS MANAGER Enter Cisco Unified CallManager Serviceability Verify Service Activation Verify CM Service Are Started Return to the Cisco Unified CallManager Administration Real Time Monitoring Tool VERIFY AVAYA AURA COMMUNICATION MANAGER CONFIGURATION CALL SCENARIOS VERIFIED ACRONYMS CONCLUSIONS REFERENCES of 71

4 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. 4 of 71

5 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 Equipment and Software Validated The following equipment and software were used for the sample configuration. Hardware Component Software Version Avaya Aura Communications Manager S8800 Server with G650 Media Gateway acting as an Evolution Server (R016x ) Patch 1002 S8810 Media Server Session Manager System Manager 6.0 ( ) Avaya 9630 IP Telephone (SIP) 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: Platform Version : Cisco 7960G IP Telephone (SIP) Phone Load: P0S Cisco 7060G IP Telephone (SCCP) Phone Load: P of 71

6 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 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 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: 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 Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: Maximum Number of DS1 Boards with Echo Cancellation: Maximum TN2501 VAL Boards: Maximum Media Gateway VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: of 71

7 AAR/ARS Routing Check Verify that ARS and ARS/AAR Dialing without FAC are enabled (on page 3 of systemparameters 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 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 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 7 of 71

8 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-SM ASM2-SM ATT-MEDPRO-1A BSM-LSP BSM1-SM CUCM IPOR MgmtPC VAL01a clan-1a clan-1a default gateway procr procr6 :: xfire-1a 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 Multiple Level Precedence & Preemption? n Multiple Locations? y Personal Station Access (PSA)? y PNC Duplication? n Port Network Support? y Posted Messages? n Private Networking? y Processor and System MSP? y Processor Ethernet? y Remote Office? y Restrict Call Forward Off Net? y Secondary Data Module? n Station and Trunk MSP? y Station as Virtual Extension? y System Management Data Transfer? n Tenant Partitioning? n Terminal Trans. Init. (TTI)? y Time of Day Routing? y TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Usage Allocation Enhancements? y Wideband Switching? n Wireless? y 8 of 71

9 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 Interregion 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: DIFFSERV/TOS PARAMETERS Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: P/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 Interregion 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: DIFFSERV/TOS PARAMETERS Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: P/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 9 of 71

10 2.4. Configure SIP Signaling Group and Trunk Group 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 10 of 71

11 Add SIP Trunk Group Add the corresponding trunk group controlled by each signaling group using the add trunkgroup 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 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 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 Note: To avoid extra SIP messages, all SIP trunks connected to Session Manager should be configured with a minimum value of 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 11 of 71

12 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 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 Multiple Level Precedence & Preemption? n Multiple Locations? y Personal Station Access (PSA)? y PNC Duplication? n Port Network Support? y Posted Messages? n Private Networking? y Processor and System MSP? y Processor Ethernet? y Remote Office? y Restrict Call Forward Off Net? y Secondary Data Module? n Station and Trunk MSP? y Station as Virtual Extension? y System Management Data Transfer? n Tenant Partitioning? n Terminal Trans. Init. (TTI)? y Time of Day Routing? y TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Usage Allocation Enhancements? y Wideband Switching? n Wireless? y 12 of 71

13 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 Total Administered: Maximum Entries: 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. 13 of 71

14 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. 14 of 71

15 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 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: 15 of 71

16 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: Click Commit to save. Name: The authoritative domain name Notes: Descriptive text (optional) The screen below shows the information for Communication Manager Evolution Server in the sample configuration. 16 of 71

17 The screen below shows the information for sample configuration which includes SIP domains for avaya.com and cucm.com. 17 of 71

18 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: Click Commit to save. IP Address Pattern: An IP address pattern used to identify the location Notes: Add a brief description The screen below shows the information for Communication Manager Evolution Server which is in the xxx subnet. 18 of 71

19 Add locations for all of the following: Avaya Aura Communication Manager Evolution Server (Location 1 Subnet x) Avaya Aura Session Manager & Avaya Aura System Manager (Location 1 Subnet x) Cisco Unified Communications Manager (CUCM Location, *) 19 of 71

20 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 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= 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) 20 of 71

21 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. 21 of 71

22 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 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 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. 22 of 71

23 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. 23 of 71

24 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. Click Commit to save. 24 of 71

25 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 and click on Select. Verify the selected SIP Entity displays on the Routing Policy Details page. Click on Commit to save. 25 of 71

26 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) 26 of 71

27 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). Click on Commit to save. 27 of 71

28 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 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. Notes: Add a brief description. Location: Select the location of the Evolution Server as defined in section 3.2, Location 1 Subnet x. Time Zone: Select the correct time zone. The sample configuration used America/Denver. Click Commit to save. 28 of 71

29 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, SIP Entity 2: From the drop-down, select one of the SIP Entities added in Section for the Communication Manager Evolution Server. Port: From the drop-down, select the correct port to use for the Communication Manager Evolution Server, Trusted: Check box to allow as a trusted host. Notes: Add a brief description. Click Commit to save. The following screen shows the entity link defined for the Communication Manager Evolution Server. 29 of 71

30 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. Notes: Add a brief description. 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 for the Communication Manager Evolution Server. The selected SIP Entity displays on the Routing Policy Details page. Click on Commit to save. Shown below is the updated screen for the sample configuration. 30 of 71

31 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 Notes: Add a brief description. 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 for Communication Manager Evolution Server.. Click on Commit to save. Shown below is the updated screen for the sample configuration. 31 of 71

32 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 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: Address of CUCM>:8443/ccmadmin/showHome.do Log in using the appropriate Username and Password Verify License From the administration tool bar, select SystemLicensingLicense Unit Report 32 of 71

33 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: DLUs 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] 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. 33 of 71

34 4.3. Configure Cisco Unified CallManager Select System Cisco Unified CM. Click on Find and select the CallManager (Publisher) 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. 34 of 71

35 4.4. Phone Network Time Protocol (NTP) Reference Select System Phone NTP Reference. Click on Find and select Add New. Click on Save. IP Address: Enter the IP Address of the NTP source. Description: Enter a brief description of the NTP source. Mode: Select Directed Broadcast 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 35 of 71

36 Click on Add Phone NTP References button. A popup window will appear with a list of configured NTP references. Click on Save. Select the NTP reference by checking the box next to the desired NTP and clicking on Add Selected button Region Information Select System Region. Click on Find and select the Default region 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. 36 of 71

37 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 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: 37 of 71

38 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 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. 38 of 71

39 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). Verify the Device Pool assigned is Default. 39 of 71

40 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). Verify the Device Pool assigned is Default. 40 of 71

41 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. 41 of 71

42 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). Verify the Device Pool assigned is Default. 42 of 71

43 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. 43 of 71

44 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. 44 of 71

45 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: Click on Save. Device Pool Name: Default Date/Time Group: CMLocal Media Resource Group List: MRGL_1 45 of 71

46 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 46 of 71

47 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. 47 of 71

48 Under SIP Information: Click on Save. 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 of 71

49 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 Route Option: Select the radio button, Route this pattern Call Classification: Select OffNet Provide Outside Dial Tone: Check the box 49 of 71

50 Under Calling Party Transformations: Use Calling Party s External Phone Number Mask: Calling Party Transform Mask: Calling Line ID Presentation: Calling Name Presentation: Check this box Enter 555XXXX Select Allowed Select Allowed Click on Save. 50 of 71

51 4.13. Add Phones 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 51 of 71

52 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: User Hold MOH Audio Source: Select MRGL_1 Select SampleAudioSource Network Hold MOH Audio Source: Select SampleAudioSource Location: Select Hub_None 52 of 71

53 53 of 71

54 Under Protocol Specific Information: Click on Save. Device Security Profile: Select Cisco 7960 Standard SIP Non-Secure Profile SIP Profile: Select Standard SIP Profile 54 of 71

55 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. 55 of 71

56 Under Directory Number Settings: User Hold MOH Audio Source: Network Hold MOH Audio Source: Select SampleAudioSource 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 A 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 56 of 71

57 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. 57 of 71

58 Add Line 2 DN Add a second DN with number 8011 to Line 2 of the phone configured in Section Follow the same steps in Section for the Line 2 DN configuration using the appropriate values. The sample configuration for Line 2 DN is shown below: 58 of 71

59 Repeat steps in Section for second 7960 SIP Phone Phone Name: Phone A2 SIP Line 1 DN: 8002 Line 2 DN: Add 7960 SCCP Phone Add the 7960 SCCP phone using the same steps described in Section using the phone type of SCCP. The remaining configuration steps for both the phone and DNs will remain the same described Section 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: of 71

60 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. 60 of 71

61 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 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 61 of 71

62 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. 62 of 71

63 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 63 of 71

64 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 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. 64 of 71

65 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 65 of 71