configuring sip trunks between avaya communication manager

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Avaya Solution & Interoperability Test Lab

Configuring SIP Trunks between Avaya Communication Manager and the AudioCodes Mediant 5000 Media Gateway – Issue 1.0

Abstract

The AudioCodes Mediant 5000 Media Gateway allows customers to consolidate facilities and reduce communications costs by concentrating PSTN traffic over DS3 lines and routing the calls over SIP trunks to Avaya Communication Manager. These Application Notes describe the administration for the Mediant 5000 Media Gateway and Avaya Communication Manager running on an Avaya S8720 Media Server with G650 Media Gateway, and are a result of early deployment testing performed by the Solution and Interoperability Test Lab.





1. Introduction The AudioCodes Mediant 5000 Media Gateway provides a means for customers to consolidate facilities and reduce communications costs by concentrating PSTN traffic over DS3 facilities. For high call traffic applications such as call centers, using DS3 facilities can provide a higher density, lower cost solution compared with DS1 facilities. The Mediant 5000 is a carrier class product that supports up to 8000 channels of SIP VoIP telephony. It uses N+1 redundancy of media gateway, Ethernet switch, shelf controller, and power supply modules to achieve high availability in mission critical applications. In the sample call center configuration shown in Figure 1, a simulated PSTN delivers customer calls using a DS3 interface to the Mediant 5000, which converts the calls to SIP and routes them to Avaya Communication Manager running on an Avaya S8720 Media Server with Avaya G650 Media Gateway. Avaya Communication Manager delivers the calls to agent phones. Agents can also make outbound calls using the same network. Note that a SIP trunk connects the Mediant 5000 and Avaya Communication Manager directly – an Avaya SIP Enablement Services server is not required. In the sample configuration, multiple C-LANs are configured to support alternate routing as well as to permit load balancing of calls delivered by the Mediant 5000. The Mediant 5000 is shipped with an Element Management System (EMS) that is used for operations, administration, management, and provisioning functions. A Solaris based EMS server communicates with the Mediant 5000 Gateway using SNMP. An EMS client communicates with the EMS server from a Microsoft Windows based PC, and provides the graphical user interface. These Application Notes focus on the administrative steps required for configuring:

• A SIP trunk between the Mediant 5000 and Avaya Communication Manager • Inbound and outbound call routing • Load balancing of inbound traffic among multiple C-LANs • Alternate routing of inbound traffic when C-LANs become inaccessible • SNMP trap receivers to which the Mediant 5000 will report alarms

Administration for Avaya Communication Manager should apply to Avaya S8500 and S8700- series Media Servers. Refer to the appropriate Avaya product documentation for general telephone, server, and call center administration details. It is assumed that the AudioCodes Mediant 5000, EMS server, and EMS client have already been installed.





Figure 1: Sample Configuration using the AudioCodes Mediant 5000 Media Gateway

2. Equipment and Software Validated The equipment and software used for the sample configuration are shown below in Table 1. Equipment Software Avaya S8720 Media Server with G650 Media Gateway

Avaya Communication Manager 3.1.2, Patch 12249

TN799DP Control LAN Interface (C-LAN) 17 TN2602AP IP Media Resource 320 221 TN2312BP IP Server Interface 31 AudiCodes Mediant 5000 3.2.67 AudiCodes EMS client 3.2.110 AudiCodes EMS server 3.2.110

Table 1: Equipment and Software Versions

3. Configure Avaya Communication Manager The following sections describe how to configure SIP trunks and call routing in Avaya Communication Manager for the sample configuration. Other aspects of SIP administration can be found in Reference [1].

3.1. Configure SIP trunks This section describes how to configure the SIP trunks that communicate with the Mediant 5000. The following commands are to be executed at the System Access Terminal (SAT).





3.1.1. Set Applicable System Parameters Use the display system-parameters customer-options command to verify that an adequate number of SIP trunk members are administered for the system. One trunk group member is consumed for each call. In the sample configuration, an IP Media Resource 320 board is used, so this must also be reflected on Page 2. On Page 3, verify that ARS is enabled. display system-parameters customer-options Page 2 of 10 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 5 0 Maximum Concurrently Registered IP Stations: 200 1 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: 5 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 5 0 Maximum Administered SIP Trunks: 300 0 Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: 0 0 Maximum G250/G350/G700 VAL Sources: 1 1 Maximum TN2602 Boards with 80 VoIP Channels: 0 0 Maximum TN2602 Boards with 320 VoIP Channels: 1 0 Maximum Number of Expanded Meet-me Conference Ports: 0 0 (NOTE: You must logoff & login to effect the permission changes.)





display system-parameters customer-options Page 3 of 10 OPTIONAL FEATURES Abbreviated Dialing Enhanced List? n Audible Message Waiting? n Access Security Gateway (ASG)? n Authorization Codes? n Analog Trunk Incoming Call ID? n Backup Cluster Automatic Takeover? n A/D Grp/Sys List Dialing Start at 01? n CAS Branch? n Answer Supervision by Call Classifier? n CAS Main? n ARS? y Change COR by FAC? n ARS/AAR Partitioning? y Computer Telephony Adjunct Links? n ARS/AAR Dialing without FAC? n Cvg Of Calls Redirected Off-net? n ASAI Link Core Capabilities? n DCS (Basic)? n ASAI Link Plus Capabilities? n DCS Call Coverage? n Async. Transfer Mode (ATM) PNC? n DCS with Rerouting? n Async. Transfer Mode (ATM) Trunking? n ATM WAN Spare Processor? n Digital Loss Plan Modification? n ATMS? n DS1 MSP? y Attendant Vectoring? n DS1 Echo Cancellation? n (NOTE: You must logoff & login to effect the permission changes.)

3.1.2. Define Codec Sets Define the audio codec sets to be used for calls that will be using the Mediant 5000. In the sample configuration, all calls will use G.711 mu-law. Codec set 1 will be defined for this purpose. Use the change ip-codec-set command to define the codec set. Default values can be used for Page 2. 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 2 20 2: 3:

3.1.3. Define IP Network Regions Define IP network region 1 using the change ip-network-region command. On Page 1, set the Authoritative Domain to the desired domain. The Name field can be set to a meaningful value. Enter the Codec Set and indicate whether inter- and intra- region media shuffling should be enabled.





change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: companyx.com Name: Company X 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? y UDP Port Max: 40001 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5





3.1.4. Define IP Node Names Define the IP addresses of the two C-LANs and of the Mediant 5000 Eth1 interface (see Section 4.2.2, Step 2) using the change node-names ip command. change node-names ip Page 1 of 1 IP NODE NAMES Name IP Address Name IP Address clan1 10 .1 .2 .213 . . . clan2 10 .1 .2 .214 . . . medpro1 10 .1 .2 .215 . . . m5000 10 .1 .2 .63 . . . default 0 .0 .0 .0 . . .

3.1.5. Add C-LAN Interfaces Use the add ip-interface command to configure each C-LAN in the Avaya G650 Media Gateway. In the sample configuration, C-LANs have been installed in slots 1a02 and 1a031. Fill in the fields as shown in bold. The IP address will be filled in automatically based on the Node Name entered. add ip-interface 1a02 Page 1 of 1 IP INTERFACES Type: C-LAN Slot: 01A02 Code/Suffix: TN799 D Node Name: clan1 IP Address: 10 .1 .2 .213 Subnet Mask: 255.255.255.0 Link: Gateway Address: 10 .1 .2 .1 Enable Ethernet Port? y Allow H.323 Endpoints? y Network Region: 1 Allow H.248 Gateways? y VLAN: n Gatekeeper Priority: 5 Target socket load and Warning level: 400 Receive Buffer TCP Window Size: 8320 ETHERNET OPTIONS Auto? y

1 For best performance, it is recommended that 2 C-LANs be configured for each DS3 facility configured in the Mediant 5000.





Use the add data-module command to complete administration for the C-LAN. Enter Ethernet for the Type, and an appropriate Name. Enter the C-LAN slot and port 17 for Port. Enter a Link number that is unique for each C-LAN. add data-module 20000 Page 1 of 1 DATA MODULE Data Extension: 20000 Name: CLAN 1 Type: ethernet Port: 01A0217 Link: 1 Network uses 1's for Broadcast Addresses? y Repeat the steps in this section for the second C-LAN in slot 1a03 with node name clan2.

3.1.6. Add MEDPRO Interface Use the add ip-interface command to configure the IP Media Resource 320 board in the Avaya G650 Media Gateway. In the sample configuration, it is installed in slot 1a04. Fill in the fields as shown in bold. The IP address will be filled in automatically based on the Node Name entered. add ip-interface 1a04 Page 1 of 1 IP INTERFACES Critical Reliable Bearer? n Type: MEDPRO Slot: 01A04 Code/Suffix: TN2602 Node Name: medpro1 IP Address: 10 .1 .2 .215 Subnet Mask: 255.255.255.0 Gateway Address: 10 .1 .2 .1 Enable Ethernet Port? y Network Region: 1 VLAN: n VOIP Channels: 320 ETHERNET OPTIONS Auto? y





3.1.7. Define SIP trunk Use the add signaling-group command to add a signaling group for one of the SIP trunks to the Mediant 5000, and fill in the indicated fields. Near-end Node Name should be set to the node name for C-LAN 1, and Far-end Node Name should be set to the node name for the Eth1 interface on the Mediant 5000, as defined in Section 3.1.4. The Far-end Domain should be the same domain entered in the IP network region form. Default values can be used for the remaining fields. add signaling-group 2 Page 1 of 1 SIGNALING GROUP Group Number: 2 Group Type: sip Transport Method: tls Near-end Node Name: clan1 Far-end Node Name: m5000 Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: 1 Far-end Domain: companyx.com Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y IP Audio Hairpinning? n Session Establishment Timer(min): 120 Add the corresponding trunk group controlled by this signaling group via the add trunk-group command. Be sure to enter the signaling group number in Signaling Group and the number of trunk members desired in Number of Members. The sample configuration uses 5 DS1’s out of the total of 28 in a single DS3 interface, so 120 members will be adequate. add trunk-group 2 Page 1 of 21 TRUNK GROUP Group Number: 2 Group Type: sip CDR Reports: y Group Name: M5000 COR: 1 TN: 1 TAC: 102 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 2 Number of Members: 120





Default values can be used for Page 2. On Page 3, set Numbering Format to be public. Subsequent pages need not be changed. The trunk members will be automatically generated based on the value of Number of Members on Page 1. add trunk-group 2 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public Prepend '+' to Calling Number? n Replace Unavailable Numbers? n Similarly, define the signaling and trunk groups for the SIP trunk from the second C-LAN to the Mediant 5000, as shown below. The far end IP address for both SIP trunks is the address of the active TP6310 board in the Mediant 5000 (Eth1 in Figure 1). If the redundant TP6310 board switches into service due to a failure of the active board, the same IP address will be used. add signaling-group 3 Page 1 of 1 SIGNALING GROUP Group Number: 3 Group Type: sip Transport Method: tls Near-end Node Name: clan2 Far-end Node Name: m5000 Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: 1 Far-end Domain: companyx.com Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y IP Audio Hairpinning? n Session Establishment Timer(min): 120





add trunk-group 3 Page 1 of 21 TRUNK GROUP Group Number: 3 Group Type: sip CDR Reports: y Group Name: M5000 COR: 1 TN: 1 TAC: 103 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 3 Number of Members: 120 add trunk-group 3 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public Prepend '+' to Calling Number? n Replace Unavailable Numbers? n

3.1.8. Define Public Numbering Format Together with the Numbering Format field in the trunk group administration of the previous section, the change public-unknown-numbering command will enable Avaya Communication Manager to provide the proper calling number when outbound calls are placed over the SIP trunks through the Mediant 5000. The entries shown specify a full 10-digit number (CPN Prefix plus 5-digit extension) to be sent to the Mediant 5000 for calls from stations whose 5-digit extension begins with 6. change public-unknown-numbering 0 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Total Ext Ext Trk CPN CPN Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Len Code Grp(s) Prefix Len 5 6 2 73212 10 5 6 3 73212 10





3.2. Configure Call Routing

3.2.1. Inbound calls In the sample configuration, inbound calls from the Mediant 5000 will be directly mapped to 5-digit extensions in Avaya Communication Manager. The change inc-call-handling-trmt command is used for this purpose. The following two screens show how incoming calls arriving at each SIP trunk are routed to the extension specified by the last 5 digits of the called number. The first six digits are deleted if the called number is 11 digits long, and the first five are deleted if the length is 10 digits. change inc-call-handling-trmt trunk-group 2 Page 1 of 30 INCOMING CALL HANDLING TREATMENT Service/ Called Called Del Insert Feature Len Number tie 11 6 tie 10 5 change inc-call-handling-trmt trunk-group 3 Page 1 of 30 INCOMING CALL HANDLING TREATMENT Service/ Called Called Del Insert Feature Len Number tie 11 6 tie 10 5

3.2.2. Outbound Calls In the sample configuration, Automatic Route Selection (ARS) will be configured for outbound calls through the Mediant 5000. Use the change dialplan analysis command to add the single-digit feature access code for ARS dialing. change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Percent Full: 0 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 3 dac 2 5 ext 5 5 ext 6 5 ext 9 1 fac # 3 fac





Use the change feature-access-codes command to specify 9 as the access code for ARS dialing. change feature-access-codes Page 1 of 5 FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: Abbreviated Dialing List2 Access Code: Abbreviated Dialing List3 Access Code: Abbreviated Dial - Prgm Group List Access Code: Announcement Access Code: Answer Back Access Code: Attendant Access Code: Auto Alternate Routing (AAR) Access Code: Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2: Automatic Callback Activation: Deactivation: Call Forwarding Activation Busy/DA: All: Deactivation: Call Park Access Code: Call Pickup Access Code: CAS Remote Hold/Answer Hold-Unhold Access Code: CDR Account Code Access Code: Change COR Access Code: Change Coverage Access Code: Contact Closure Open Code: Close Code: Contact Closure Pulse Code: Use the change ars analysis command to configure the route selection, based on the number dialed following the “9”. In the sample configuration, outbound local calls are dialed in the form AAANNNXXXX. The entry below defines the number format to be matched and the routing pattern selected. Other area codes can be specified to use the same route pattern and trunk to the Mediant 5000 in the same manner. change ars analysis 1 Page 1 of 2 ARS DIGIT ANALYSIS TABLE Location: all Percent Full: 0 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd 732 10 10 1 fnpa n





Use the change route-pattern command to define the SIP trunk groups to be selected for the corresponding route pattern, as shown below. In the sample configuration, the SIP trunks corresponding to the C-LANs just administered are included. change route-pattern 1 Page 1 of 3 Pattern Number: 10 Pattern Name: PSTN via 3825 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: 2 0 n user 2: 3 0 n user 3: n user 4: n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR 0 1 2 3 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 none 3: y y y y y n n rest none 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 Use the change locations command to designate the SIP trunk routing pattern defined above in the Proxy Sel. Rte. Pat. field. This provides support for features such as call transfer. change locations Page 1 of 1 LOCATIONS ARS Prefix 1 Required For 10-Digit NANP Calls? y Loc. Name Timezone Rule NPA Proxy Sel. No. Offset Rte. Pat. 1: Main + 00:00 0 1

4. Configure the AudioCodes Mediant 5000 Gateway The following sections describe the steps for configuring the SIP and PSTN trunks and call routing for the Mediant 5000. Configuration is performed using the EMS client GUI-based provisioning system, which supports Microsoft Windows. It is assumed that the Mediant 5000, EMS server, and EMS client have already been installed. See References [3-5] for installation information.





4.1. Configure Network Support Parameters Steps Description 1. Invoke the GUI provisioning system from the Microsoft Windows client PC by double-clicking

the desktop icon as shown below:

2. The login screen will be displayed. Enter the login and password, and the IP address of the EMS server.





Steps Description 3. The main GUI provisioning screen will be displayed. In the Regions List pane on the right are

the logical/geographical regions where media gateways reside. Double-click on the appropriate row entry.





Steps Description 4. The media gateway list will be displayed in the MGs List pane. Select the entry corresponding to

the Mediant 5000 to be configured, and click on Properties.





Steps Description 5. To enter IP addresses of the NTP server and SNMP trap receivers, click on Network Services.

Under NTP, enter the NTP server IP address. Two SNMP trap receivers can be designated by filling in the IP address of the receiver, the Trap Port (default 162), and Security Profile (select EMSSecurityProfile(1)) under the NMS and OSS sections. Click on Apply and then Close. Note that the parameters in this step are optional, and can be configured as necessary to match the operational needs of the customer.





4.2. Configure the TP6310 Board The following sections describe the administrative steps for configuring the SIP and DS3 trunks associated with the TP6310 board, as well as the call routing rules required for the sample configuration.

4.2.1. Lock the Board The board must be locked to perform the steps in the following sections. Note that locking the board removes it from service, and is therefore disruptive. On the MGs List pane (see Step 4 in Section 4.1), double-click on the row corresponding to the Mediant 5000. The MG Status tab will be highlighted and the Status pane will be shown, depicting a replica of the front panel. Board slots are numbered from 1 to 10 from bottom to top on the left side. In the sample configuration, installed boards includes the TP6310 DS3 boards (slots 9 and 10), Ethernet switch boards (slots 3 and 4), and shelf controller boards (slots 1 and 2). Click on the active TP6310 board shown in black, and use mouse button 2 to select Maintenance->Lock. Click Yes in the confirmation window.





4.2.2. Configure Board Level Parameters Steps Description 1. In the Status pane under the MG Status tab (see previous section), select the locked TP6310

board. A locked board is indicated by a blue “locking pin” on its right hand side (see slot 9). Select the Properties link.





Steps Description 2. In the Board6310 Parameters Provisioning window displayed (shown below in abbreviated

form), click on General Settings on the left side. Set IP Address 1 to the IP address for this board (“Eth1” as shown in Figure 1). Set None Mode Clk Source to pstn and SSL/TLS Negotiation to TLSv1Only. Default values can be used for the remaining fields. Click on Apply and then Close.

3. Repeat Steps 1 and 2 for the other TP6310 board in slot 10, using the Eth2 IP address as shown in Figure 1.





Steps Description 4. Click on Voice on the left hand side. Set DSP Load File Version to 0. Set the Jitter Buffer

Min Delay to 10 milliseconds, since in the sample configuration, VoIP traffic will be on an internal enterprise network with low delay characteristics. Click on Apply and then Close.





4.2.3. Configure DS3 Trunk Steps Description 1. Double-click on the locked TP6310 board in the Status pane under the MG Status tab. The DS3

Status pane shows the status of each of the 3 DS3 interfaces on the board. Double-click on the DS3 for which the DS1 channel interface parameters are to be defined.

2. The DS1 Carriers List pane is displayed, listing the 28 DS1 interfaces comprising this DS3. To set the DS1 parameters for a particular interface, double-click on its entry in the list.





Steps Description 3. The Trunk Parameters Provisioning window is displayed. Fill in the fields as required by the

service provider. Click on Apply.





Steps Description 4. Click on the ISDN/DPNSS tab on the left, and set the Termination Side to the appropriate

value, usually userTerminationSide if the PSTN connection is to a service provider. Click on Apply.





Steps Description 5. The Profile Management pane in the Trunk Parameters Provisioning Window can be used to

define a configuration profile that can be applied to many DS1 interfaces, saving configuration steps. For instance, the configuration performed in Steps 2-4 can be saved by clicking on the Save button. In the dialogue box that is displayed, fill in a name for this profile, and click on OK.

6. Select a range of DS1 interfaces to which the profile will be applied, and right click the mouse, selecting Apply Profile.





Steps Description 7. In the Select Profile dialog box, use the pull-down menu on Select profile to enter the profile to

be applied, in this case profile1. Click on OK.





Steps Description 8. A Results Table window will be displayed indicating successful application of the profile. Click

on Done to close this window.





4.2.4. Configure SIP Trunk Steps Description 1. Double-click on the locked TP6310 board in the Status pane under the MG Status tab. Click on

the SIP tab. Then click on the Protocol Settings tab.

2. The SIP Protocol Definitions window is displayed (shown below in abbreviated form). The trunks between the Mediant 5000 and Avaya Communication Manager use Transport Layer Security (TLS)2. Set SIP Destination Port, Enable Early Media, and Transport Type to the values shown. Click on Apply and then Close.

2 The procedure for installing an Avaya TLS certificate on the Mediant 5000 is proprietary. These applications notes assume that certificate installation has been completed prior to these configuration steps.





Steps Description 3. Click on the Coders tab to set the codec preferences for SIP calls routed to Avaya

Communication Manager. Click on the “+” icon to add each codec, ordered from most to least preferred.





Steps Description 4. In the SIP Coders window that is displayed, fill in a NAME if desired, and verify the remaining

field values shown below for G.711 mu-law. These values should match those defined for Avaya Communication Manager in Section 3.1.2. Click on Apply and then Close.

5. Repeat Steps 3 and 4 for each codec. Only one codec is used in the sample configuration. The resulting codec list is shown below.





Steps Description 6. Click on the Trunk Groups tab. The Sip Trunk Group List pane will be displayed. Click on the

“+” icon to add a trunk group.





Steps Description 7. Fill in fields in the SIP Trunk Groups window that is displayed. For the sample configuration,

this trunk group is configured to include all DS1 trunks in the three DS3 interfaces on the TP6310 board, and has the Name “All trunks.” Set Trunk Group ID to 1, First Trunk ID to 1 (first DS1 in the first DS3), Last Trunk ID to 84 (last DS1 in the third DS3), Starting Channel to 1 (first DS0 in each DS1), and Ending Channel to 24 (last DS0 in each DS1). The Starting Phone Number is used when no ANI information is available in incoming PSTN calls, and should be set to an appropriate 7-digit number3. Each DS0 channel in the trunk group will be assigned a unique number based on the Starting Phone Number. Click on Apply and then Close.

3 The number must be 7 digits long for call distribution to multiple C-LANs to work properly (see Section 4.2.5, Step 3).





Steps Description 8. The Sip Trunk Group List pane will now show the added trunk group.

9. Click on the Trunk Group Settings tab. In the SIP Trunk Groups Settings window displayed, set Trunk Group ID and Channel Select Mode to appropriate values. Click on Apply and then on Close.





4.2.5. Configure Call Routing Similar to Avaya Communication Manager, the Mediant 5000 must be configured with rules to route inbound (DS3 -> SIP) and outbound (SIP -> DS3) calls based on information in the incoming call request. In the sample configuration, it is desired to distribute the incoming traffic from the Mediant 5000 across the two C-LANs in the Avaya G650 Media gateway. All outbound calls should be routed from the C-LANs to the Mediant 5000, which will route them to the DS3 interface on the PSTN side. These sample routing rules are summarized below. The steps in this section implement these rules. Call Type Routing Condition

Inbound Mediant 5000(DS3) -> Mediant 5000(SIP) -> C-LAN 1

Last digit of calling number = 0-4

Inbound Mediant 5000(DS3) -> Mediant 5000(SIP) -> C-LAN 2

Last digit of calling number = 5-9

Outbound C-LANs -> Mediant 5000(SIP) -> Mediant 5000(DS3) ALL

Steps Description 1. Double-click on the locked TP6310 board in the Status pane under the MG Status tab. Click on

the SIP tab, then on the Routing tab. The SIP Tel to IP Routing List pane is displayed. Routing rules in this list are applied to inbound calls (DS3 -> SIP). Click on the “+” icon to add a routing rule.





Steps Description 2. In the SIP Routing Tel to Ip window that is displayed, enter an appropriate Name. Enter a match

specification in Source Phone Prefix. X matches any digit in the calling number, and 0-4 specifies that the digit in that position should match any digit in the range 0 to 4. Dest Address should contain the IP address of the C-LAN to which the incoming PSTN call should be routed. In this example, if the last digit of the calling number ends in 0, 1, 2, 3, or 4, then a SIP INVITE will be sent to C-LAN 1. Click on Apply, and then Close.





Steps Description 3. Repeat the previous step, clicking on the “+” icon to add each rule. In the sample configuration,

another rule is needed for callers whose last digit ends in 5, 6, 7, 8, or 9. After adding this rule, the routing entries look as shown below. Note that these two rules implement a form of load balancing that is based on the calling numbers of the inbound traffic.

4. The Sip Tel To IP Routing List also has the property that if a call cannot be successfully routed to the destination specified by a matched rule, then the next rule in the list that matches will be used to attempt to route the call. This can be used to implement a failover strategy if one of the C-LANs is inaccessible. For the sample configuration, two additional routing rules can be added such that for each call, if the initial C-LAN (C-LAN 1 or C-LAN 2, depending on the calling number) is not available, the call will be routed to the other C-LAN. The result of adding these two rules is shown below, using the procedure outlined in Steps 1-2. The two new rules are identical to the previous two, except that the Dest Address for each points to the other C-LAN.





Steps Description 5. To enable the alternate routing behavior described in the previous step, click on the Routing

Setting tab. In the SIP Routing Properties window displayed, set Enable Alternative Routing to Yes. Defaults can be used for the other fields. Click on Apply and then Close.





Steps Description 6. Click on the IP To Tel tab. The Sip IP To Tel Routing List pane is displayed. Routing rules in

this list are applied to outbound calls (SIP->DS3). Click on the “+” icon to add a routing rule.





Steps Description 7. In the SIP Routing Ip to Tel window that is displayed, enter an appropriate Name. Enter a match

specification in Dest Phone Prefix, Source Phone Prefix, and Source Address, and the destination PSTN trunk group in Trunk Group ID. In the sample configuration, all calls are to be routed to Trunk Group 1, so a “*” is entered in the fields as shown below, specifying a match on any value. Click on Apply and then Close.





Steps Description 8. Repeat the previous step for each rule to be added. In the sample configuration, only one rule is

needed. After adding this rule, the routing entries look as shown below. When finished adding rules, click on the icon to return to the Status pane under the MG Status tab.

4.2.6. Unlock the Board The board must be unlocked for the above configuration to be applied to the TP6310 board, after which it is reset and enabled for service. Select the locked TP6310 board and right-click the mouse to select Maintenance->Unlock. Click Yes in the confirmation window. The TP6310 board will reset and return to service after several minutes. The Alarm Browser pane at the bottom of the window will indicate the status of the board.





5. Verification Steps The following steps can be used to troubleshoot network configurations in the field: Network connectivity: 1. For the SIP network, ping the Eth1 interface on the Mediant 5000 from the C-LANs in the

Avaya G650 Media Gateway. 2. Using the status signaling-group command, verify that the SIP trunks are up between the

Avaya Communication Manager and the Mediant 5000, and that the transport protocols and ports match.

3. For the PSTN side, verify that the DS3 and DS1 trunks are up on the Mediant 5000 by verifying that the icons for those entries on the DS3 Status and DS1 Carriers List panes are green (see Section 4.2.3. Steps 1-2). If not, check cabling and also check with the service provider as to what the proper settings should be.

Call routing: 1. Verify successful inbound and outbound calls between Avaya Communication Manager and

the Mediant 5000. If inbound (PSTN -> SIP) calls fail, verify that the proper Tel to IP routing rules have been defined in the Mediant 5000 to support routing SIP calls to the appropriate C-LANs. If outbound calls fail, verify that the proper IP to Tel rules have been defined to route calls to the appropriate DS3/DS1, and use list trace station on Avaya Communication Manager for the calling phone to verify that the call is being properly routed through the SIP trunk to the Mediant 5000.

2. If SIP signaling issues are suspected, change the transport protocol to TCP on the SIP trunk and use a SIP-capable protocol analyzer to monitor the signaling messages. On Avaya Communication Manager, change the Transport Method to tcp and Far-end Listen Port to 5060 (Section 3.1.7). On the Mediant 5000, change the Transport Type and SIP Destination Port to the same values (Section 4.2.4).

3. In general, if a SIP device in the network does not receive an expected response to a SIP message it has transmitted, the standard procedure is to retransmit that message at an exponentially increasing interval (½ second, then 1 second, etc.), up to six more times. Look for this behavior at various points in the SIP call routing path if calls are not successful. Most likely, the expected message has not been routed properly through the IP network.

6. Conclusion These Application Notes have provided the administrative details on configuring SIP trunks and call routing between Avaya Communication Manager and the AudioCodes Mediant 5000. Configuration of SNMP trap receivers on the Mediant 5000 was also covered. This information should be applicable to other Avaya Media Servers and Media Gateways running Avaya Communication Manager.





7. Additional References [1] SIP Support in Release 3.1 of Communication Manager Running on the S8300, S8400,

S8500, S8500B, S8700, and S8710 Media Server, Issue 1.4, Doc ID 555-245-206, February, 2006, available at http://support.avaya.com.

[2] Administrator Guide for Avaya Communication Manager, Issue 2, Doc ID 03-300509, February 2006, available at http://support.avaya.com.

[3] AudioCodes EMS User’s Manual, Version 3.2, Document # LTRT-91007, Sections 3-5, available at http://www.audiocodes.com.

[4] Element Management System (EMS) Server Installation, Operation & Maintenance Manual, Version 3.2, Document # LTRT-94109, Sections 12, 13.5, 13.7-13.9, available at http://www.audiocodes.com.

[5] IPmedia 5000 Installation, Operation, Maintenance, Version 3.2, Document # LTRT-89602, Sections 4-9, 11, available at http://www.audiocodes.com.

Field Code Changed

Deleted: [6] SIP Support in Release 3.0 of Communication Manager Running on the S8300, S8500, S8700, and S8710 Media Server, Issue 5.1, Doc ID 555-245-206, July, 2005, available at http://support.avaya.com.¶[7] Administrator Guide for Avaya Communication Manager, Issue 1, Doc ID 03-300509, June 2005, available at http://support.avaya.com.





1. Introduction............................................................................................................................. 2 2. Equipment and Software Validated ........................................................................................ 3 3. Configure Avaya Communication Manager........................................................................... 3

3.1. Configure SIP trunks....................................................................................................... 3 3.1.1. Set Applicable System Parameters ......................................................................... 4 3.1.2. Define Codec Sets................................................................................................... 5 3.1.3. Define IP Network Regions .................................................................................... 5 3.1.4. Define IP Node Names ........................................................................................... 7 3.1.5. Add C-LAN Interfaces............................................................................................ 7 3.1.6. Add MEDPRO Interface......................................................................................... 8 3.1.7. Define SIP trunk ..................................................................................................... 9 3.1.8. Define Public Numbering Format......................................................................... 11

3.2. Configure Call Routing................................................................................................. 12 3.2.1. Inbound calls......................................................................................................... 12 3.2.2. Outbound Calls ..................................................................................................... 12

4. Configure the AudioCodes Mediant 5000 Gateway............................................................. 14 4.1. Configure Network Support Parameters....................................................................... 15 4.2. Configure the TP6310 Board ........................................................................................ 19

4.2.1. Lock the Board...................................................................................................... 19 4.2.2. Configure Board Level Parameters....................................................................... 20 4.2.3. Configure DS3 Trunk ........................................................................................... 23 4.2.4. Configure SIP Trunk............................................................................................. 29 4.2.5. Configure Call Routing......................................................................................... 35 4.2.6. Unlock the Board .................................................................................................. 41

5. Verification Steps.................................................................................................................. 42 6. Conclusion ............................................................................................................................ 42 7. Additional References........................................................................................................... 43





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