Configuring multiple Avaya Aura Session Managers Release 5.2 to address different Network Failure Scenarios Issue 1.0

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1 Avaya Solution Interoperability Test Lab Configuring multiple Avaya Aura Session Managers Release 5.2 to address different Network Failure Scenarios Issue 1.0 Abstract These Application Notes describe a sample configuration with two Avaya Aura TM Session Managers Release 5.2 configured as an active-active redundant pair. Two Session Managers are used so that one Session Manager can act as backup for the other in case of network or a Session Manager failure. The network in the sample configuration uses SIP trunks between the Avaya Aura TM Session Managers, Avaya Business Communication Manager Release 5.0, an Avaya Aura TM Communication Manager Release operating as an Access Element, and a second Avaya Aura TM Communication Manager operating as a Feature Server. These Application Notes provide 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 TM Communication Manager Access Element Verify System Capabilities and Licensing Verify SIP Trunk Capacity Verify AAR/ARS Routing Configure Trunk-to-Trunk Transfers Configure Codec Type Set IP Network Region Add Node Names and IP Addresses Configure SIP Signaling Group and Trunk Group Add Signaling Groups for SIP Trunks Add SIP Trunk Groups Configure Route Pattern Administer Numbering Plan Administer Uniform Dialplan Administer AAR Analysis Save Translations Configure Avaya Aura TM Communication Manager Feature Server Verify System Capabilities and Licensing SIP Trunk Capacity Check AAR/ARS Routing Check Enable Private Networking Configure Trunk-to-Trunk Transfers Add Node Name of Avaya Aura TM Session Manager Configure IP Network Region Configure SIP Signaling Groups and Trunk Groups Add Signaling Groups for SIP Trunks Add SIP Trunk Groups Configure Route Patterns Administer Numbering Plan Administer Private Numbering Plan Administer AAR Analysis of 71

3 3.7. Configure Stations Configure Off-PBX-Telephone Station-Mapping Save Translations Configure Avaya Aura Session Manager Administer SIP Domains Define Locations Configure each Avaya Aura TM Session Manager Define SIP Entity Configure Listen Ports Add each Avaya Aura TM Session Manager to System Manager Define an Entity Link between Avaya Aura TM Session Managers Add Avaya Business Communication Manager Define SIP Entity Define Entity Links Define Routing Policy Define Dial Pattern Add Avaya Aura TM Communication Manager Access Element Define Local Host Name Resolution Define SIP Entity Define Entity Links Define Routing Policy Define Dial Pattern Add Avaya Aura TM Communication Manager Feature Server Define a SIP Entity and Entity Link Define Routing Policy Configure Avaya Business Communication Manager Open Applications Web Page Download Avaya Business Element Manager Add Avaya Business Communication Manager as an Element Navigation Add SIP Trunk to each Avaya Aura TM Session Manager Configure Routing Table for SIP Trunk Configure IP Trunk Settings Configure SIP Settings of 71

4 Configure SIP Proxy Configure SIP Media Parameters Define Business Name Configure Dial Plan Configure SIP Line Pools Configure Public Network Configure Routing Configure Destination Code Verification Steps Verify Avaya Aura TM Session Manager Configuration Verify each Avaya Aura TM Session Manager is operational Verify SIP Link Status Verify Registrations of SIP Endpoints Verify Avaya Business Communication Manager Configuration Verify Avaya Aura TM Communication Manager Call Scenarios Verified Issues Found and Known Limitations Acronyms Conclusions Additional References of 71

5 1. Introduction These Application Notes describe a sample configuration with two Avaya Aura TM Session Managers Release 5.2 configured as an active-active redundant pair. Two Session Managers are used so that one Session Manager can act as backup for the other in case of network or a Session Manager failure. In addition, the network in the sample configuration uses SIP trunks between the Session Managers, Avaya Business Communication Manager Release 5.0, an Avaya Aura TM Communication Manager Release operating as an Access Element, and a second Avaya Aura TM Communication Manager, operating as a Feature Server. As shown in Figure 1, Avaya 9600 Series IP Telephone (H.323) and 2420 Digital Telephone are supported by Communication Manager Access Element. The Communication Manager Access Element is connected over SIP trunks to both of the Session Managers, using the SM-100 (Security Module) network interface on each Session Manager. The Avaya Business Communication Manager Release 5.0 runs on the Business Communication Manager 50 platform and supports the 1230 UNIstim IP and T7631E Digital Telephone. The Business Communication Manager is also connected to both Session Managers over SIP trunks, using the SM-100 (Security Module) network interface on each Session Manager. All intersystem calls are carried over these SIP trunks IP Telephones configured as SIP endpoints utilize the Session Manager User Registration feature and require a Communication Manager operating as a Feature Server. Communication Manager Feature Server only supports IMS-SIP users that are registered to Session Manager. The Communication Manager Feature Server is connected to one of the Session Managers via an IMS-enabled SIP signaling group and associated SIP trunk group. All SIP users managed by the Communication Manager Feature Server should be configured to register to the same Session Manager. Session Manager is managed by Avaya Aura System Manager. For the sample configuration, two Session Managers running on separate Avaya S8510 Servers are deployed as a pair of active-active redundant servers. The Communication Manager Access Element runs on a pair of duplicated Avaya S8730 Servers with Avaya G650 Media Gateway. The Communication Manager Feature Server runs on an Avaya S8300 Server. The results in these Application Notes should be applicable to other Avaya servers and media gateways that support Communication Manager These Application Notes will focus on the configuration of the SIP trunks and call routing. Detailed administration of Communication Manager Feature Server, SIP endpoints, or SIP users will not be described (see the appropriate documentation listed in Section 9). 5 of 71

6 Figure 1 Sample Configuration 6 of 71

7 1.1. Equipment and Software Validated The following equipment and software were used for the sample configuration. Note: S8300C was used during the setup and testing of this configuration, however S8300D and S8800 are officially supported servers for the Avaya Aura TM Communication Manager Feature Server. Component Software Version Avaya Aura TM Session Manager on Avaya S8510 Server Release Avaya Aura TM System Manager Release 5.2, Load: Avaya Aura TM Communication Manager Access Element Release 5.2.1, SP1 Duplicated Avaya S8730 Servers Load: R015x Avaya G650 Media Gateway Avaya Aura TM Communication Manager Feature Server Avaya S8300 Server Avaya IP Telephones: 4621SW 9620 Release 5.2.1, SP1 Load: R015x FW: 2.90 FW:3.0 Avaya SIP Telephones 9630 FW: Avaya Digital Telephones (2420D) N/A Avaya Business Communication Manager Release 5, Version: Update: BCM050.R500.SU.System UNIstim IP Telephone FW: 062AC6R T7316E Digital Telephone N/A 7 of 71

8 2. Configure Avaya Aura TM Communication Manager Access Element This section describes the administration of Communication Manager Access Element using a System Access Terminal (SAT). Some administration screens have been abbreviated for clarity. Verify System Capabilities and Communication Manager Licensing Administer Codec Set Administer IP Network Region Administer IP Node Names Administer SIP Trunk Group and Signaling Group Administer Route Pattern Administer Numbering Plan After completing these steps, the save translation 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 Verify SIP Trunk Capacity Issue the display system-parameters customer-options command to verify that an adequate number of 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: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: 0 0 Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: of 71

9 Verify AAR/ARS Routing To simplify the dialing plan for users of SIP endpoints, verify that both the ARS and ARS/AAR Dialing without FAC parameters are enabled on Page 3 of the system-parameters customeroptions command. display system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES A/D Grp/Sys List Dialing Start at 01? n 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? Configure Trunk-to-Trunk Transfers Use the change system-parameters features command to enable trunk-to-trunk transfers. This feature is needed to transfer a call from remote switch to either the same or another switch. The Trunk-to-Trunk Transfer field was set to all to enable all trunk-to-trunk transfers on a system wide basis. Note that this feature poses significant security risk by increasing the risk of toll fraud, and must be used with caution. To minimize the risk, a COS could be defined to allow trunk-to-trunk transfers for a specific trunk group(s). For more information regarding how to configure a Communication Manager to minimize toll fraud, see reference in Section 9. change system-parameters features Page 1 of 18 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 9 of 71

10 2.2. Configure Codec Type Issue the change ip-codec-set n command where n is the number used to identify the codec set. Enter the following values: Audio Codec: Enter G.711MU and G.729 as supported types of Audio Codecs Silence Suppression: Retain the default value n. Frames Per Pkt: Enter 2. Packet Size (ms): Enter 20. Media Encryption: Enter the value based on the system requirement. For the sample configuration, none was used. change ip-codec-set 1 Page 1 of 2 IP Codec Set Codec Set: 1 Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : G.729 n : Media Encryption 1: none 2.3. Set IP Network Region Using the change ip-network-region 1 command, enter the following values: Intra-region IP-IP Direct Audio: Enter yes Inter-region IP-IP Direct Audio: Enter yes. Codec Set: Enter the corresponding audio codec set configured in Section 2.2. Authoritative Domain: Enter correct SIP domain for the configuration. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: Authoritative Domain: avaya.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: of 71

11 2.4. Add Node Names and IP Addresses Using the change node-names ip command, add the node-name and IP Addresses for the CLANs and the SM-100 interfaces for both Session Managers, if not previously added. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address ASM ASM CLAN CLAN of 71

12 2.5. Configure SIP Signaling Group and Trunk Group Add Signaling Groups for SIP Trunks Use the add signaling-group n command, where n is an available signaling group number to create the SIP signaling groups to connect to both Session Managers. In order to test different types of network failures such a network failure associated with one of the CLANs, four SIP Signaling Groups and associated Trunk Groups were defined between the Access Element and each Session Manager. In the sample configuration, trunk groups 10 and 15 and signaling groups 10 and 15 were used to connect to the first Session Manager. Trunk groups 20 and 25 and signaling groups 20 and 25 were used to connect to the second Session Manager. For each signaling group, fill in the indicated fields as shown below on the screen that shows the values for one of the signaling groups. Default values can be used for the remaining fields. Note: TCP was used for the sample configuration, however TLS would be typically used in production environment. Group Type: Enter sip Transport Method: Enter tcp IMS Enabled: Enter n Near-end Node Name: Enter one of the C-LAN node names from Section 2.4. Far-end Node Name: Enter one of the Session Manager node names from Section 2.4. Near-end Listen Port: Enter 5060 Far-end Listen Port: Enter 5060 Far-end Domain: Enter Authoritative Domain configured in Section 2.3 DTMF over IP: rtp-payload Session Establishment Timer: 3 add signaling-group 10 Page 1 of 1 SIGNALING GROUP Group Number: 10 IMS Enabled? n IP Video? n Group Type: sip Transport Method: tcp Near-end Node Name: CLAN-2 Far-end Node Name: ASM1 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: Far-end Domain: avaya.com Bypass If IP Threshold Exceeded? 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? n Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6 12 of 71

13 The following screen shows the set of Signaling Groups defined for the sample configuration. list signaling-group SIGNALING GROUPS Grp Group Trunk Pri D-Ch/ Sec D-Ch/ Max Max No. Adm'd No. Type FAS? Brds Near-node Far-node NCA TSC CA TSC NCA TSCs 1 isdn-pri y 1 01A isdn-pri y 1 01A sip y CLAN-1 ASM1 11 sip y CLAN-1 mx1 15 sip y CLAN-2 ASM1 20 sip y CLAN-1 ASM2 25 sip y CLAN-2 ASM2 13 of 71

14 Add SIP Trunk Groups Add the corresponding trunk groups 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. In the sample configuration, trunk groups 10 and 15 and signaling groups 10 and 15 were used to connect to the first Session Manager. Trunk groups 20 and 25 and signaling groups 20 and 25 were used to connect to the second Session Manager. For each trunk group, fill in the indicated fields as shown below. Default values can be used for the remaining fields Group Type: Enter sip Group Name: Enter a descriptive name. TAC: Enter an available trunk access code. Service Type: Enter tie Signaling Group: Enter the number of the signaling group added in Section Number of Members: Enter the number of members in the SIP trunk to be allocated to calls routed to Session Manager (must be within the limits of the total number of trunks configured in Section 2.1.1). Once the add command is completed, trunk members will be automatically generated based on the value in the Number of Members field. The screen below shows the values for one of the trunk groups defined in the sample configuration. add trunk-group 10 Page 1 of 21 TRUNK GROUP Group Number: 10 Group Type: sip CDR Reports: y Group Name: SIP trunk to ASM1 COR: 1 TN: 1 TAC: #10 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 10 Number of Members: of 71

15 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 add 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 On Page 3, set the Numbering Format field to public. Use default values for all other fields. add trunk-group 10 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public UUI Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Show ANSWERED BY on Display? y On Page 4, set the Mark Users As Phone field to y to send correct user information to Business Communication Manager in the SIP messages, and verify the Telephone Event Payload Type field is set to 120. add trunk-group 10 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? y Prepend '+' to Calling Number? n Send Transferring Party Information? n Network Call Redirection? n Send Diversion Header? n Support Request History? n Telephone Event Payload Type: of 71

16 The following screen shows the set of Trunk Groups defined for the sample configuration. list trunk-group TRUNK GROUPS Grp No. Out Que No. TAC Group Type Group Name Mem TN COR CDR Meas Dsp Len 1 #01 isdn PSTN Trunk y none y 0 2 #02 isdn QSIG to Nortel C1000e y none y 0 10 #10 sip SIP trunk to ASM y none n 0 11 #11 sip OUTSIDE CALL y none n 0 15 #15 sip SIP Trunk to ASM y none y 0 20 #20 sip SIP trunk to ASM y none n 0 25 #25 sip SIP Trunk to ASM y none y Configure Route Pattern This section provides the configuration of the route patterns used in the sample configuration for routing calls between Communication Manager Access Element, Business Communication Manager, and SIP users registered to Session Manager. Use the add route-pattern n command, when n is an available number to define a route pattern for routing calls over any of the SIP trunk groups defined in Section to either Session Manager. In the Look Ahead (LAR) field for each trunk group, select next to enable Access Element to use alternative trunk groups in the case of network failures. In the sample configuration, route pattern 10 was created as shown below. add route-pattern 10 Page 1 of 3 Pattern Number: 10 Pattern Name: SIP to both ASM SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 10 0 n user 2: 15 0 n user 3: 20 0 n user 4: 25 0 n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest next 2: y y y y y n n rest next 3: y y y y y n n rest next 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none 16 of 71

17 2.7. Administer Numbering Plan Administer Uniform Dialplan To enable stations on the Communication Manager Access Element to call SIP phones registered to Session Manager, add an entry for extension numbers associated with SIP phones to the uniform dial plan. Use the change uniform-dialplan n command, where n is the first digit of the extension numbers used for SIP stations. In the sample configuration, extensions starting with 666-3XXX are used for extensions associated with the 9630 SIP phones. Note: The dial plan shown below is an example dial plan that was used in the sample configuration. Other dial plans may be appropriate for different customer networks. change uniform-dialplan 6 Page 1 of 2 UNIFORM DIAL PLAN TABLE Percent Full: 0 Matching Insert Node Pattern Len Del Digits Net Conv Num aar n aar n aar n aar n n Administer AAR Analysis This section provides the configuration of the AAR pattern used in the sample configuration for routing calls between Communication Manager Access Element, Business Communication Manager and SIP users registered to Session Manager. Other routing methods may be used. Use the change aar analysis n command where n is the first digit of number used to route calls to stations on Business Communication Manager. In the sample configuration, all calls starting with 333 will be routed to Business Communication Manager using Route Pattern 10 and all calls starting with 6663 will be routed to SIP stations also using Route Pattern 10. change aar analysis 3 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd aar n aar n aar n aar n aar n aar n 17 of 71

18 2.8. Save Translations Configuration of Communication Manager Access Element is complete. Use the save translation command to save these changes. 3. Configure Avaya Aura TM Communication Manager Feature Server This section describes the administration of the SIP trunks between the Communication Manager Feature Server and one of the Session Managers using a System Access Terminal (SAT). These instructions assume the G450 Media Gateway is already configured on the Communication Manager Feature Server. Some administration screens have been abbreviated for clarity. Other administrative screens are not shown in this section, as the screens are the same as screens described in Section 2. Verify System Capabilities and Communication Manager Licensing Administer IP Node Names Administer Network Region Administer SIP Trunk Group and Signaling Group Administer Route Patterns Administer Numbering Plan Administer Stations Administer Off-PBX-Telephone Station-Mapping After completing these steps, the save translation command should be performed Verify System Capabilities and Licensing This section describes the procedures to configure the correct system capabilities and licensing on the Communication Manager Feature Server. 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 Use the display system-parameters customer-options command to verify that an adequate number of SIP trunk members are administered for the system. Navigate to Page 2, and verify that there is sufficient remaining capacity for SIP trunks by comparing the Maximum Administered SIP Trunks field value with the corresponding value in the USED column. The difference between the two values needs to be greater than or equal to the desired number of simultaneous SIP trunk connections. Screen example can be viewed in Section AAR/ARS Routing Check Verify that ARS and ARS/AAR Dialing without FAC parameters are enabled on Page 3 of the system-parameters customer-options form as explained in Section of 71

19 Enable Private Networking SIP Users registered to Session Manager should be added to either private or public numbering table on the Communication Manager Feature Server. For the sample configuration, private numbering was used and all extension numbers were unique within the private network. However, in many customer networks it may not be possible to define unique extension numbers for all users within the private network. For these types of networks additional administration may be required as described in References in Section 9. Use the change system-parameters customer-options command to verify that Private Networking parameter 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? n Posted Messages? n Private Networking? y Processor and System MSP? y Processor Ethernet? y 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? n TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Usage Allocation Enhancements? y Wideband Switching? n Wireless? y Configure Trunk-to-Trunk Transfers Use the change system-parameters features command to enable trunk-to-trunk transfers, as explained in Section of 71

20 3.2. Add Node Name of Avaya Aura TM Session Manager Using the change node-names ip command, add the node-name and IP Address of the SM-100 network interface for one of the Session Managers where the SIP endpoints will be registered, if not already added. Note: Although there are two Session Managers in the sample configuration, a Communication Manager Feature Server can only be connected to one of the Session Managers. All SIP users managed by the Feature Server should also be configured to register to the same Session Manager. For the sample configuration, SIP endpoints were registered to the first Session Manager, named ASM1 with IP address change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address ASM Nortel-CS1000e default procr Configure IP Network Region Using the change ip-network-region 1 command, enter the following values: Intra-region IP-IP Direct Audio Enter yes Inter-region IP-IP Direct Audio Enter yes Authoritative Domain Enter correct SIP domain for the configuration. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: avaya.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: of 71

21 3.4. Configure SIP Signaling Groups and Trunk Groups Add Signaling Groups for SIP Trunks Use the add signaling-group n command, where n is an available signaling group number to create a SIP signaling group to connect to one of the Avaya Aura Session Managers. In the sample configuration, signaling group 10 and trunk group 10 were used to connect to the first Avaya Aura Session Manager. The screen below shows the values used for the signaling group in the sample configuration: Group Type: Enter sip Transport Method: Enter tcp IMS Enabled?: Enter y Near-end Node Name: Enter procr node name from Section 3.2 Far-end Node Name: Enter Session Manager node name from Section 3.2 Near-end Listen Port: Enter 5060 Far-end Listen Port: Enter 5060 Far-end Domain: Enter Authoritative Domain from Section 3.3 Enable Layer 3 Test: Enter y Session Establishment Timer: Enter 3 Default values can be used for the remaining fields display signaling-group 10 Page 1 of 1 SIGNALING GROUP Group Number: 10 IMS Enabled? y IP Video? n Group Type: sip Transport Method: tcp Near-end Node Name: procr Far-end Node Name: ASM1 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 Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 1 21 of 71

22 Add SIP Trunk Groups Add the corresponding trunk groups controlled by the signaling group using the add trunkgroup n command, where n is an available trunk group number and fill in the indicated fields. Group Type: Group Name: TAC: Service Type: Signaling Group: Number of Members: Enter sip Enter a descriptive name Enter an available trunk access code Enter tie Enter the number of the signaling group added in Section Enter the number of SIP trunks to be allocated to calls routed to Session Manager (must be within the limits of the total number of trunks configured in Section 3.1.1) add trunk-group 10 Page 1 of 21 TRUNK GROUP Group Number: 10 Group Type: sip CDR Reports: y Group Name: ASM1 COR: 1 TN: 1 TAC: #10 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 10 Number of Members: 10 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 add 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): of 71

23 On Page 3, set Numbering Format field to private. Use default values for all other fields. add 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? On Page 4, set Mark Users As Phone field to y to send correct user information to Business Communication Manager in the SIP messages, and verify the Telephone Event Payload Type field is set to 120. add trunk-group 10 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? y Prepend '+' to Calling Number? n Send Transferring Party Information? n Network Call Redirection? n Send Diversion Header? n Support Request History? n Telephone Event Payload Type: Configure Route Patterns This section provides the configuration of the route patterns used in the sample configuration for routing calls between Communication Manager Feature Server, Business Communication Manager and Communication Manager Access Element. Note that other methods of routing may be used. Use the add route-pattern n command, when n is an available number to define the route pattern for routing calls over the SIP trunk group defined in Section In the sample configuration, route pattern 10 was created as shown below. add route-pattern 10 Page 1 of 3 Pattern Number: 10 Pattern Name: SIP to ASM1 SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 10 0 n user 2: n user 3: n user 23 of 71

24 3.6. Administer Numbering Plan Administer Private Numbering Plan To enable SIP endpoints to dial extensions defined in the Communication Manager Access Element, SIP Users registered to Session Manager should be added to either private or public numbering table on the Communication Manager Feature Server. For the sample configuration, private numbering was used and all extension numbers were unique within the private network. However, in many customer networks it may not be possible to define unique extension numbers for all users within the private network. For these types of networks, additional administration may be required as described in References in Section 9. Use the change private-numbering n command, where n is the number used to identify the private number plan. For the sample configuration, extension numbers starting with 5XX- XXXX or 6XX-XXX are used on the Communication Manager Access Element. Ext Len: Ext Code: Trunk Grp: Private Prefix: Enter the extension length allowed by the dial plan Enter leading digit (s) from extension number Enter the SIP Trunk Group number for the SIP trunk between the Feature Server and Session Manager 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: of 71

25 Administer AAR Analysis This section provides the configuration of the AAR pattern used in the sample configuration for routing calls between Communication Manager Feature Server, Communication Manager Access Element and Business Communication Manager. Note that other methods of routing may be used. Use the change aar analysis n command where n is the first digit of the number used to route calls to stations on Business Communication Manager. In the sample configuration, all calls starting with 333 will be routed to Business Communication Manager using Route Pattern 10 and all calls starting with 666 will be routed to the Access Element also using Route Pattern 10. change aar analysis 3 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd aar n aar n aar n aar n aar n aar n 3.7. Configure Stations For each SIP user defined in Session Manager, the corresponding station on the Communication Manager Feature Server can either be manually configured or automatically generated when a new SIP user is defined in System Manager. For more information on manually configuring stations or automatically generating a station when adding a new SIP user, see References in Section of 71

26 3.8. Configure Off-PBX-Telephone Station-Mapping Use the change off-pbx-telephone station-mapping command for each extension associated with SIP users defined in Session Manager. On Page 1, enter the SIP Trunk Group defined in Section in the Trunk Selection field and use default values for other fields. change off-pbx-telephone station-mapping Page 1 of 3 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Application Dial CC Phone Number Trunk Config Dual Extension Prefix Selection Set Mode OPS On Page 2, enter the following values: Mapping Mode: Enter both Calls Allowed: Enter all change off-pbx-telephone station-mapping Page 2 of 3 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Appl Call Mapping Calls Bridged Location Extension Name Limit Mode Allowed Calls OPS 3 both all none Save Translations Configuration of Communication Manager Feature Server is complete. Use the save translation command to save these changes Note: After a change on the Communication Manager Feature Server which alters the dial plan, synchronization between Communication Manager Feature Server and Session Manager needs to be completed and SIP phones must be re-registered. To request an on demand synchronization, log into the System Manager console and use the Synchronize CM Data feature under the Communication System Management menu. 26 of 71

27 4. Configure Avaya Aura Session Manager This section provides the procedures for configuring two Session Managers as an active-active redundant pair and includes the following items: Administer SIP domain Define Logical/Physical Locations For each Session Manager in the sample configuration: o Define SIP Entity o Configure the Listen Port for TCP and UDP connections o Add each Session Manager to System Manager o Define an Entity Link between the two Session Managers to support failover scenarios For the Business Communication Manager and Communication Manager Access Element: o Define the corresponding SIP Entity o Define Entity Links which define the SIP trunk parameters used by Session Manager when routing calls to/from SIP Entities o Define Routing Policies which control call routing between the SIP Entities o Define Dial Patterns which govern to which SIP Entity a call is routed Define the Communication Manager Feature Server as an administration entity Add SIP Endpoints/SIP URE users Configuration is accomplished by accessing the browser-based GUI of System Manager, using the URL where <ip-address> is the IP address of System Manager. Log in with the appropriate credentials and accept the Copyright Notice. 27 of 71

28 4.1. Administer SIP Domains Expand Network Routing Policy link on the left side of the navigation menu and select SIP Domains. Click New and enter following values: Name Enter the Authoritative Domain Name specified for the Communication Manager Access Element in Section 2.3. Notes Enter a brief description. [Optional] Click Commit to save. The screen below shows the name of the SIP domain used in the sample configuration. 28 of 71

29 4.2. Define Locations Expand Network Routing Policy and select Locations. Locations are used to identify logical and/or physical locations where SIP Entities reside, for purposes of bandwidth management or location-based routing. Click New and specify following values: In the General section, Name Enter a descriptive name Notes Enter a brief description. [Optional] In the Location Pattern section click Add. In the IP Address Pattern field enter the pattern used to logically identify the location. Click Commit to save. The screen below shows the location information for Communication Manager Access Element in the sample configuration. 29 of 71

30 4.3. Configure each Avaya Aura TM Session Manager Define SIP Entity Verify a SIP entity has been defined for each Session Manager. Expand Network Routing Policy and select SIP Entities. Click New to define a SIP Entity for the first Session Manager, or if previously defined, select the first Session Manager and click Edit. In the General Section enter the following values: Name Enter a descriptive name FQDN or IP Address Enter the Host Name or IP address of the SM100-Interface for the first Session Manager Location Select Location defined in Section 4.2 from dropdown menu Notes Enter a brief description [Optional] Outbound Proxy Leave blank Defaults can be used for other fields. Click Commit to save. The screen below shows the SIP Entity details for the first Session Manager in the sample configuration. Repeat the above process to define a SIP Entity for the second Session Manager. 30 of 71

31 Configure Listen Ports Session Manager has the ability to translate communication between two SIP entities that use different ports and protocols. For each type of protocol that will be used in the network, add a corresponding listen port to Session Manager. Since Avaya Business Communication Manager 5.0 only supports UDP connections, a listen port for UDP connections should be configured on each Session Manager. Expand Network Routing Policy and select SIP Entities. Select one of the Session Managers and Click Edit. In the Port Section click Add and enter following values: Port Enter 5060 Note: Session Manager is able to use the same port for both TCP and UDP connections. Protocol Select UDP from the drop-down menu Default Domain Select the domain name defined in Section 4.1 Note: Do not select ALL from the drop-down menu as the value for Default Domain field. Instead, the Default Domain for the listen port must be configured to use the domain name defined in Section 4.1. Notes Enter a brief description. [Optional] Click Commit to save. The following screen shows the result after adding a listen port for UDP connections using port 5060 for one of the Session Managers. Repeat the process to add a listen port for TCP connections if not already added. Repeat the above process to configure the listen port for UDP connections on the second Session Manager. 31 of 71

32 Add each Avaya Aura TM Session Manager to System Manager To enable System Manager to manage each Session Manager, the network connection between System Manager and each Session Manager needs to be defined. Expand Session Manager link and select Session Manager Administration. Select New to add the first Session Manager to System Manager. In the General Section, enter following values: SIP Entity Name Select the first Session Manager defined in Section from the drop-down menu Management Access Point Host Name/IP Enter the IP address of the server running Session Manager (not the IP address of the SM-100 module) In the Security Module section, enter following values: Network Mask Enter the network mask corresponding to the IP address of Session Manager server Default Gateway Enter the IP address of the default gateway Click Commit to save. The screen below shows the System Management details for the first Session Manager in the sample configuration. Repeat the above process to define a network connection between System Manager and the second Session Manager. 32 of 71

33 Define an Entity Link between Avaya Aura TM Session Managers To support failover scenarios, an entity link should be defined between the two Session Managers. Expand Network Routing Policy and select Entity Link. Click New and enter the following values: Name Enter an identifier for the failover link. SIP Entity 1 Select the first Session Manager from the drop-down menu SIP Entity 2 Select the second Session Manager from the drop-down menu Protocol Select TCP from drop-down menu Port Enter 5060 to match default listen port defined in Section Notes Enter a brief description [Optional] Click Commit to save. The following screen shows the entity link defined between the two Session Manager instances: 33 of 71

34 4.4. Add Avaya Business Communication Manager The following section captures the relevant screens for configuring Avaya Business Communication Manager applicable for the sample configuration Define SIP Entity Expand Network Routing Policy and select SIP Entities. Click New. In the General Section, enter the following values: Name Enter a name for the Business Communication Manager. FQDN or IP Address Enter the IP address of the Business Communication Manager. Type Select Other from the drop-down menu. Notes Enter a brief description. [Optional] Location Select the Location added in Section 4.2 from the drop-down menu. Note: Selecting a value for the Location field is optional unless location-based routing is being used. Click Commit to save. The following screen shows addition of Business Communication Manager. 34 of 71

35 Define Entity Links Expand Network Routing Policy and select Entity Link. Click New. Enter the following values: Name Enter a name for the link to the Business Communication Manager. SIP Entity 1 Select the first Session Manager from the drop-down menu SIP Entity 2 Select the SIP Entity added for the Business Communication Manager in Section from the drop-down menu. Protocol Select UDP from drop-down menu. Port Enter 5060 to match the listen port defined for UDP connections in Section Notes Enter a brief description [Optional] Click Commit to save. The following screen shows the entity link defined between the Business Communication Manager and the first Session Manager. Repeat the above steps to define an Entity Link between Business Communication Manager and the second Session Manager. The following screen shows both of the entity links defined in the sample configuration for the Business Communication Manager. 35 of 71

36 Define Routing Policy Expand Network Routing Policy and select Routing Policies. Click New. In the General Section, enter the following values: Name Enter a name to identify the routing policy for the Business Communication Manager Notes Enter a brief description. [Optional] In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not shown). Select the entry of the Business Communication Manager added in Section and click Select. The selected SIP Entity displays on the Routing Policy Details page. Click Commit to save. The following screen shows the routing policy defined for routing calls to the Business Communication Manager. Note: The routing policy defined in this section is an example and was used in the sample configuration. Other routing policies may be appropriate for different customer networks. 36 of 71

37 Define Dial Pattern Expand Network Routing Policy and select Dial Patterns. Click New. Under General section, enter the following values: Pattern Enter the dial pattern for any extension numbers associated with stations on the Business Communication Manager. Min Enter the minimum number of digits that must be dialed. Max Enter the maximum number of digits that may be dialed. SIP Domain Select All to indicate Session Manager should accept incoming calls from all SIP domains. Notes Enter a brief description.[optional] Under Locations and Routing Policies section, click Add (not shown). The Locations and Routing Policy List page opens. Locations Select the desired location. Routing Policies Name Select the Routing Policy defined for Business Communication Manager in Section Click Select. Click Commit to save. The following screen shows the dial pattern defined for routing calls to the Business Communication Manager in the sample configuration. 37 of 71

38 4.5. Add Avaya Aura TM Communication Manager Access Element The following section captures the relevant screens for configuring Communication Manager Access Element on the System Manager, applicable for the sample configuration Define Local Host Name Resolution Since there will be multiple SIP entity links between the Communication Manager Access Element and each Session Manager, a FQDN should be defined for the Communication Manager Access Element to enable Session Manager to resolve to multiple IP addresses for this SIP Entity. Expand Network Configuration under Session Manager. Select Local Host Name Resolution. Click New. Enter the following values: Host Name (FQDN) Enter a FQDN for Communication Manager Access Element IP address Enter the IP address for one of the CLAN boards on Communication Manager Access Element. Repeat the above steps to add an entry for the second CLAN board on the Access Element. Click Commit to save. The following screen shows the Local Host Name Resolution defined for the Communication Manager Access Element in the sample configuration. 38 of 71

39 Define SIP Entity Expand Network Routing Policy and select SIP Entities. Click New. In the General Section, enter the following values: Name Enter an identifier for the Communication Manager Access Element. FQDN or IP Address Enter the FQDN defined for the Communication Manager Access Element in Section Type Select CM from the drop-down menu. Location Select the Location added in Section 4.2 from the drop-down menu. Notes Enter a brief description. [Optional] Click Commit to save. The following screen shows the SIP entity defined for the Communication Manager Access Element. 39 of 71

40 Define Entity Links Define an Entity Link between the Communication Manager Access Element and each Session Manager to support failover, in case of network or a Session Manager failure. Expand Network Routing Policy and select Entity Links. Click New. Enter the following values: Name Enter an identifier for the Access Element. SIP Entity 1 Select the first Session Manager from the drop-down menu. SIP Entity 2 Select the SIP Entity added in Section Trusted Check the checkbox Protocol Select TCP from drop-down menu. Notes Enter a brief description. [Optional] Click Commit to save. The following screen shows the entity link defined between the Communication Manager Access Element and the first Session Manager. Repeat the above process to define an Entity Link between the Communication Manager Access Element and the second Session Manager. The following screen shows both of the Entity Links defined for the Communication Manager Access Element in the sample configuration. 40 of 71

41 Define Routing Policy Expand Network Routing Policy and select Routing Policies. Click New. In the General section, enter the following values: Name Enter an identifier to define the routing policy for the Communication Manager Access Element Notes Enter a brief description. [Optional] In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not shown). Name Select entry for the Communication Manager Access Element added in Section and click Select. The selected SIP Entity displays on the Routing Policy Details page. Click Commit to save. The following screen shows the Routing Policy defined for the Communication Manager Access Element. 41 of 71

42 Define Dial Pattern Expand Network Routing Policy and select Dial Patterns. Click New. In the General section, enter the following values: Pattern Enter the dial pattern for any extension numbers associated with stations on the Communication Manager Access Element Min Enter the minimum number of digits that must be dialed. Max Enter the maximum number of digits that may be dialed. SIP Domain Select the SIP Domain added in Section 4.1 or select All from the dropdown menu, if Session Manager is able to accept incoming calls from all SIP domains. Notes Enter a brief description. [Optional] In the Locations and Routing Policies section, click Add. The Locations and Routing Policy List page opens (not shown). Locations Select the desired location. Routing Policy Name Select the policy defined for Communication Manager Access Element in Section Click Select. Click Commit to save. Note: The Dial Pattern defined in this section is an example and was used in the sample configuration. Other dial patterns and routing policies may be appropriate for different customer networks. 42 of 71

43 4.6. Add Avaya Aura TM Communication Manager Feature Server The following section captures relevant screens for configuring the Communication Manager Feature Server to enable registered SIP users to make or receive calls from stations on the Business Communication Manager or from the Communication Manager Access Element. In addition to the steps described in this section, other administration activities will be needed such as defining an Application Sequence for the Feature Server or adding new SIP users. For more information on these additional administration activities, see References in Section 9. Note: Although there are two Session Managers in the sample configuration, a Communication Manager Feature Server can only be connected to one of the Session Managers. All SIP users managed by the Feature Server should also be configured to register to the same Session Manager Define a SIP Entity and Entity Link Follow the steps described in Sections and to add a SIP Entity and Entity Link for the Communication Manager Feature Server. The following screen shows addition of Communication Manager Feature Server and associated entity link to one of the Session Managers for the sample configuration. The IP address used is that of the S8300C server. 43 of 71

44 Define Routing Policy Since the SIP users are registered on Session Manager, the Routing Policy defined for the Communication Manager Feature Server does not need to include any dial patterns. The following screen shows the Routing Policy defined for the Communication Manager Feature Server with no Dial Patterns. 44 of 71

45 5. Configure Avaya Business Communication Manager This section describes the relevant configuration of the SIP Trunks and call routing between the Business Communication Manager and each Session Manager. In addition to the steps described in this section, other administration activities will be needed such as verifying licensing, configuring digital and IP stations, and defining the associated target lines for each station on the Business Communication Manager. For more information on these additional administration activities, see References in Section 9. The Business Communication Manager is configured using the Business Element Manager GUI Open Applications Web Page During the installation and initial configuration phase, the installation technician should first connect to the OAM IP port on the Business Communication Manager Server. After the system is configured with the appropriate IP settings for the customer LAN, the URL to open the Applications web page will be the IP address of the Business Communication Manager Server. See references and product installation documentation in Section 9 for more information on initial installation procedures including configuring IP settings. Enter appropriate User name and Password in the Authentication dialog box as shown below. 45 of 71

46 After successful login, the following Welcome screen will be displayed Download Avaya Business Element Manager Select the Business Element Manager from the Applications list (as shown on the previous screen) and select Run to download the application to the desktop. Wait for several seconds while the Business Element Manager application downloads. Enter the default user name and password to log into the Business Element Manager. Select Confirm to acknowledge the copyright notice. 46 of 71

47 5.3. Add Avaya Business Communication Manager as an Element After successful login to Business Element Manager, right click the Network Elements folder in the Element Navigation Panel as shown below. Select New Network Element from the first pop-up menu (not shown) and select Business Communication Manager from the second pop-up menu (not shown). Enter IP address for the Business Communication Manager server and the default User ID and Password in the Business Communication Manager Entry dialog as shown below and select OK. 47 of 71

48 Details of the Business Communication Manager server are displayed, as shown below. Finally, to connect to the Business Communication Manager server, enter the default Password in the screen shown above and select the Connect button in the Toolbar Navigation After connecting to the server, the initial Business Element Manager screen appears. If the Element Manager GUI is being used to configure a single Business Communication Manager server, then click the arrow in the upper right corner of the Element Navigation Panel to hide this panel. 48 of 71

49 The screen below shows Business Element Manager GUI displayed without the Element Navigation Panel. Use the Task Navigation Panel to navigate to specific configuration tasks Add SIP Trunk to each Avaya Aura TM Session Manager Navigate to Resources Telephony Resources task in the Task Navigation Panel. Select the IP Trunks row in the Telephony Resources table. Wait for the configuration details of IP Trunks to be displayed in the lower section of the screen as shown below. 49 of 71

50 Configure Routing Table for SIP Trunk Routing Table is the first tab that appears in configuration details for IP trunks. Since Business Communication Manager is configured to connect to both Session Managers, an entry in SIP Trunk Routing Table is not necessary. Instead, see Section for more information on configuring the SIP Proxy settings Configure IP Trunk Settings Under the IP Trunk Settings tab, verify Send name display, Remote capability MWI and Forward redirected OLI are checked as shown below Configure SIP Settings Under the SIP Settings tab, In Telephony Settings section, select Enabled-All from the drop-down menu to re-route calls over PSTN line if SIP trunk fails. In the SIP Settings section, enter the following values: o Call signaling port: Verify the number matches the port number configured for the Business Communication Manager SIP Entity Link defined in Section o Local Domain: Leave field blank. o Disable maddr in Contact: Check the checkbox. o Disable OPTIONS Caps: Check the checkbox In the RFC2833 section, enter the same payload number in Dynamic Payload field as defined for the Telephone Event Payload Type field on Page 4 of the Trunk Group screen for the SIP Trunk Groups on Communication Manager (see Sections and 3.4.2). For the sample configuration, the default value of 120 was used. Default values can be used for other fields. 50 of 71

51 Configure SIP Proxy Under the SIP Proxy tab, configure the SIP Trunk connections to both Session Managers. In the SIP Proxy section on the left side of the page configure the following fields, Domain: Enter the same SIP Domain name as defined in Section 4.1. Route all calls using proxy: Check this option to route all external calls to one of the Session Managers MCDN Protocol: Select None from the drop-down menu. In the Outbound Proxy Table on the bottom of the page, select the Add button to add a row to configure a SIP trunk for each Session Manager. Name: Enter a name for SIP Trunk IP Address: Enter the IP Address of the SM-100 card for the Session Manager Port: Enter the UDP port number to which Business Communication Manager will send SIP messages. This value should match the value defined for UDP connections on Session Manager in Section Load-Balancing Weight: Enter a larger value for the primary Session Manager. Keep Alive: Select OPTIONS from the drop-down menu. Business Communication Manager uses the response from SIP OPTIONS message to determine if the first Session Manager is available. If it determines the first Session Manager is not available, it automatically routes calls to the second Session Manager. The following dialog shows the values entered for the first Session Manager. The following screen shows the Outbound Proxy Table settings for the sample configuration. 51 of 71

52 Configure SIP Media Parameters Under the SIP Media Parameters tab, configure the Business Communication Manager to use the same set of Codecs as defined for the Communication Manager Access Element in Section 2.2. In the Preferred Codecs section on the left side of the page, Select G.711-uLaw, G.729 from the Available List and click Add button to move these two codec choices to the Selected List table. Configure the G.711-uLaw codec as the first choice by moving G.711-ulaw to the top of list. In the Settings section on the right side of the page, Uncheck Enable Voice Activity Detection. Select 20ms as the payload size from the drop-down menu for both G.729 and G.711 Select T.38 from the drop-down menu for the Fax transport field. The following screen shows the SIP Media Parameters settings for the sample configuration. 52 of 71

53 5.6. Define Business Name Navigate to the Telephony Global Settings Features Settings task. On the Feature Settings page enter a name into the Business Name field. This name will be sent as part of the user information in SIP messages. If the Business Name field is left blank, Business Communication Manager will not include the station name in the SIP message. Note: Since Business Communication Manager concatenates the station name to the end of the Business Name in the SIP message and there appears to be a fixed length for this concatenated string, using a short Business Name is recommended. The following screen shows the Feature Settings for the sample configuration. 53 of 71

54 5.7. Configure Dial Plan Configure SIP Line Pools Navigate to the Telephony Dialing Plan Line Pools task. Select BlocA from the Line Pools table. In the Details section for BlocA, select Add button to allow each station to access the SIP trunk. Note: The BlocA Line Pool is automatically configured as a VoIP Trunk Type in Business Communication Manager. The screen below shows results for the sample configuration. 54 of 71

55 Configure Public Network Navigate to the Telephony Dialing Plan Public Network task. In the Public Network Settings section, enter the number of digits for received calls in Public Received number length parameter. In the sample configuration received calls contain 6 digits. In Public Network DN Lengths section, select the Add button to define the dialed number pattern for outgoing calls to Session Manager. In the sample configuration, originating calls routed to Session Manager will start with the digits 666 as shown in the dialog below. Click OK to enter the new prefix. Note: The dialed number pattern shown in this section is an example and was used in the sample configuration. Other dialed number patterns may be appropriate for different customer networks. Select the new row in the Public Network DN Lengths table to modify the DN Length field as shown below. Move the cursor to save the change. This change may take several seconds to finish. 55 of 71

56 Configure Routing Navigate to the Telephony Dialing Plan Routing task. Under the Routes tab, select Add button to create a route for routing calls to Session Manager. Enter an available route number in the Add Route dialog as shown below. Click OK to add the route. Select the row associated with the new route in the Routes table and select BlocA from the drop-down menu in the Use Pool column. Note: This change may take several seconds to complete as shown below. 56 of 71

57 After the Use Pool change completes under Routes tab, select Public from drop-down menu in DN Type column. The following screen shows the routes defined for the sample configuration. 57 of 71

58 Configure Destination Code Navigate to the Telephony Dialing Plan Routing task. Under the Destination Code tab, select Add button to create a destination code for routing calls to Session Manager. Enter the first digit of the number used for outgoing calls to Session Manager in the Add Destination Code dialog, as shown below. Click OK to add the destination code. Select the row associated with the new destination code in the Destination Codes table to configure the Normal Route and Absorbed Length fields. Normal Route Enter the route number defined in Section Absorbed Length Select 0 from the drop-down menu since the number used for the destination code is the first digit in the outgoing number. The following screen shows the details of the Destination Codes entry for the sample configuration. 58 of 71

59 6. Verification Steps 6.1. Verify Avaya Aura TM Session Manager Configuration Verify each Avaya Aura TM Session Manager is operational Expand the Session Manager menu on the left navigational menu and click System Status System State Administration. Verify Management State is Management Enabled and the Service State is Accept New Service for each Session Manager as shown below. Click System Status Security Module Status and verify the status of the Security Module Deployment (SM 100 network interface) is Up as shown below. 59 of 71

60 Click System Status Data Replication Status and verify the data replication is operational as shown below Verify SIP Link Status Expand the Session Manager menu on the left and click System Status SIP Entity Monitoring. Verify all SIP Entity Links are operational as shown below. 60 of 71

61 Select SIP Entity for the Business Communication Manager, BCM-50, and verify at least one of the Entity links is Up as shown below Verify Registrations of SIP Endpoints Expand User Management on left navigation menu. Verify SIP Users have been created in the Session Manager. In the sample configuration, two SIP Users were created as shown in the highlighted area below. 61 of 71

62 Verify SIP endpoints have successfully registered with the Session Manager as shown below. 62 of 71

63 6.2. Verify Avaya Business Communication Manager Configuration Use the Business Communication Monitor tool to monitor the status of SIP trunk calls. To access the tool, use the Business Communication Manager Application web page to launch the Monitor tool. Login with the same username and password used to log into the Element Manager. Navigate to the Line Monitor tab to see the status of SIP trunks. The following screen shows 3 active calls between Business Communication Manager and stations on Communication Manager Access Element. The fourth call on Line 008 is a call to a SIP station registered to Session Manager. Use IP Devices tab to monitor individual IP stations. For example, the screen below provides status of an active SIP call to station of 71

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