Configuring SIP IP Telephony Using Avaya SIP Enablement Services, Avaya Communication Manager, and the Cisco AS5400 Universal Gateway - Issue 1.

Transcription

1 Avaya Solution & Interoperability Test Lab Configuring SIP IP Telephony Using Avaya SIP Enablement Services, Avaya Communication Manager, and the Cisco AS5400 Universal Gateway - Issue 1.0 Abstract With the introduction of the SIP standard that supports telephony as well as other communication modes, there is a much broader range of SIP telephones and gateways available to customers. There will be sales opportunities involving customers who wish to purchase the Avaya SIP offer, but have a significant investment in SIP/PSTN gateways other than those offered by Avaya. Customers may be interested in replacing their existing telephony infrastructure with Avaya servers, but wish to re-use the existing non-avaya gateways. These Application Notes describe the configuration steps required to connect the Cisco AS5400 Universal Gateway to a SIP infrastructure consisting of an Avaya SIP Enablement Services (SES) server and an Avaya S8300 Server with G700 Media Gateway running Avaya Communication Manager. Two SIP trunking configurations are covered: direct SIP trunking with Avaya Communication Manager, and SIP trunking with Avaya Communication Manager via Avaya SIP Enablement Services. The configuration steps described are also applicable to other Linux-based Avaya Servers and Media Gateways running Avaya Communication Manager. 1 of 25

2 1. Introduction With the introduction of the SIP standard that supports telephony as well as other communication modes, there is a much broader range of SIP telephones and gateways available to customers. There will be sales opportunities involving customers who wish to purchase the Avaya SIP offer, but have a significant investment in SIP/PSTN gateways other than those offered by Avaya. Customers may be interested in replacing their existing telephony infrastructure with Avaya servers, but wish to re-use the existing non-avaya gateways. These Application Notes describe the configuration steps for using the Cisco AS5400 Universal Gateway with Avaya SIP Enablement Services (SES), Avaya S8300 Server, and Avaya G700 Media Gateway. Only the configuration steps pertinent to interoperability of Cisco and Avaya equipment are covered. General administration information can be found in the product documentation as well as the specific references listed in Section 10. The configuration described should be applicable to other Linux-based Avaya Servers and Media Gateways running Avaya Communication Manager Configuration The configuration used as an example in these Application Notes is shown in Figure 1. The diagram indicates logical signaling connections. With the exception of the Avaya 6408D Digital Telephone and the fax machines, all components are physically connected to a single Avaya C363T-PWR Converged Stackable Switch and are administered on a single subnet. The Avaya IA770 INTUITY AUDIX Messaging Application resides on the Avaya S8300 Server and is used to support voice messaging. The Cisco AS5400 Universal Gateway is connected to a simulated PSTN network using a T1/PRI interface. The configuration steps described in the following sections cover the routing of calls by Avaya SES 1, Avaya Communication Manager, and the AS5400. The AS5400 is configured as a peer gateway that routes calls between its T1/PRI and SIP interfaces. The SIP trunk to the Cisco AS5400 is shown in two alternate configurations: a direct SIP trunk between the Cisco AS5400 and Avaya Communication Manager, and a SIP trunk to one of the Avaya SES home servers. These will be referred to as direct and SES trunks, respectively. Table 1 profiles the network management capabilities of the AS5400. Administration mechanisms Command line interface Administration levels Craft, Administrator File transfer server TFTP Error logs Standard logs supported by Cisco IOS SNMP support Yes Table 1: Network Management Capabilities of Cisco AS5400 Universal Gateway 1 The sample configuration uses multiple SIP Enablement Services servers for illustrative purposes. For installations less than 6000 users, a single server configured as an Edge/Home combination would suffice. 2 of 25

3 Figure 1: Avaya SIP Test Configuration with Cisco AS5400 Universal Gateway 2. Equipment and Software Validated The following equipment and software were used in the configuration. Equipment Software Avaya C363T-PWR Converged Stackable Switch Avaya SIP Enablement Services (SES) Server Avaya S8300 Server with G700 Media Gateway Avaya Communication Manager (Load 415.1), Patch Avaya 6408 Digital Telephone - Avaya 4621 IP Telephone (SIP) Avaya 9630 IP Telephone (H.323) S1.5 Avaya 9640 IP Telephone (SIP) (2) Avaya one-x Desktop Edition 2.1 Service Pack 2 (Build 78) Cisco AS5400 Universal Gateway IOS 12.4 (15)T7 Table 2: Equipment and Software Versions Used 3 of 25

4 3. Supported Features Table 3 gives a summary of the features supported by the configuration steps described in these Application Notes. The features apply to both direct and SES SIP trunks. Feature Comments Extension-to-extension calls Basic calls to legacy phones Digital, H.323, Analog Caller ID DTMF RFC 2833 Fax (G.711 voice mode) Fax (T.38) Call Hold, Resume Attended Transfer Unattended Transfer Conference Compressed codecs G.729A 2 Table 3: SIP Telephony Feature Support 4. Call Routing Requirements To support outbound and inbound calls between enterprise extensions and the PSTN via the Cisco AS5400 Universal Gateway, the sample configuration implements the following call routing: 1. SES trunk calls from enterprise telephones to the PSTN must be routed by Avaya Communication Manager through the SIP trunk to Avaya SES, which must in turn route the calls to the AS5400. Calls from the PSTN to enterprise telephones must be routed by the AS5400 to Avaya SES. 2. Direct SIP trunk - Avaya Communication Manager routes the calls using the SIP trunk to the Cisco AS5400. Calls from the PSTN to enterprise telephones must be routed by the AS5400 to Avaya Communication Manager. The following sections address these requirements. 2 Avaya 4600 Series IP Telephones (SIP) not supported. All other IP telephone models supported. 4 of 25

5 5. Configure Avaya Communication Manager This section focuses on the commands required to route calls between Avaya-supported telephones and the PSTN via the Cisco AS5400. For complete documentation on SIP administration, see References [2-3]. Use the System Access Terminal (SAT) interface to perform the following steps. Log in with the appropriate credentials Verify SIP Trunk Capacity Verify that an adequate number of SIP trunk members are licensed for the system using the display system-parameters customer-options command. Navigate to Page 2 and check the Maximum Administered SIP Trunks. Note that one trunk group member is consumed for each SIP endpoint on a call, so that the maximum number of simultaneous SIP-SIP calls supported is equal to half the total number of members (two members per call), whereas SIP- H.323 or SIP-digital calls only consume one trunk member per call. If more trunks are required than are licensed for the system, contact an Avaya sales representative to obtain licensing to accommodate the additional capacity. display system-parameters customer-options Page 2 of 10 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: 5 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 5 0 Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: 0 0 Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: 0 0 Maximum Media Gateway VAL Sources: 5 1 Maximum TN2602 Boards with 80 VoIP Channels: 0 0 Maximum TN2602 Boards with 320 VoIP Channels: 0 0 Maximum Number of Expanded Meet-me Conference Ports: 0 0 (NOTE: You must logoff & login to effect the permission changes.) 5 of 25

6 5.2. Define Dial Plan Avaya Communication Manager applies dial plan and routing rules to determine where a call should be routed. In the sample configuration, Automatic Route Selection (ARS) will be configured to route calls to the PSTN via the Cisco AS5400. Use the display dialplan analysis command to make sure that the dial plan allows for a Feature Access Code (FAC) for ARS. In the sample configuration, the ARS FAC is 9. 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 0 3 fac 1 3 fac 3 5 ext 4 5 ext 5 5 ext 7 5 ext 9 1 fac 5.3. Define Feature Access Code (FAC) for ARS Use change feature-access-codes to define the access codes for ARS, shown in bold. change feature-access-codes Page 1 of 8 FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: 601 Abbreviated Dialing List2 Access Code: 602 Abbreviated Dialing List3 Access Code: 603 Abbreviated Dial - Prgm Group List Access Code: Announcement Access Code: 606 Answer Back Access Code: 605 Attendant Access Code: Auto Alternate Routing (AAR) Access Code: 8 Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2: Automatic Callback Activation: *5 Deactivation: #5 Call Forwarding Activation Busy/DA: *2 All: 612 Deactivation: #2 Call Forwarding Enhanced Status: Act: Deactivation: Call Park Access Code: 604 Call Pickup Access Code: *6 CAS Remote Hold/Answer Hold-Unhold Access Code: #6 CDR Account Code Access Code: Change COR Access Code: Change Coverage Access Code: Contact Closure Open Code: Close Code: 6 of 25

7 5.4. Define SIP Signaling Group The following sections describe the signaling group configuration for the two alternative SIP trunk arrangements SES Trunk For configurations that use a SIP trunk between Avaya SES and the Cisco AS5400, a SIP trunk must be configured between Avaya Communication Manager and Avaya SES. Outbound calls will use ARS to select this SIP trunk, based on the dialed number. Avaya SES will then use an address map to route the call to the AS5400 (see Section 6, Steps 3-9). Incoming calls select the SIP signaling group whose Far-end Domain field matches the SIP domain in the From header of the INVITE message. If no match is found, a signaling group with a blank Far-end Domain field will be selected. For this reason, it is considered best practice to configure a signaling group with a blank Far-end Domain field in order to have at least one trunk that can accept an unknown foreign domain (see Reference [3]). If such a signaling group is not already defined, use the add signaling-group command to add it. This signaling group will be used for inbound and outbound calls to and from the AS5400 via Avaya SES. Set the Near-end Node Name to procr, the IP address of the S8300 Server, and Far-end Node Name to the IP address of Avaya SES (Home 2 in the sample configuration). The addresses are defined using the change node-names ip command. Remaining fields can be left at their defaults. Note that the Far-end Domain field has been left blank, so that incoming calls from the AS5400 will select this SIP signaling group. 3 add signaling-group 15 Page 1 of 1 SIGNALING GROUP Group Number: 15 Group Type: sip Transport Method: tls Near-end Node Name: procr Far-end Node Name: Home2 Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: Far-end Domain: Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y IP Audio Hairpinning? n Enable Layer 3 Test? n Session Establishment Timer(min): 3 Alternate Route Timer(sec): 6 3 Note however that if there is more than one signaling group with a blank Far-end Domain and the same Near-end and Far-end Node Name, incoming calls will select the lowest number signaling group. 7 of 25

8 Direct Trunk For configurations that use a direct SIP trunk between Avaya Communication Manager and the Cisco AS5400, define a signaling group as in the previous section, with the following differences: 1. Far-end Node Name should correspond to the IP address of the Ethernet interface on the Cisco AS Far-end Domain should contain the IP address of the Ethernet interface on the Cisco AS Define SIP Trunk Group Use the add trunk-group command to specify the SIP trunk group corresponding to the previously defined signaling group. Set the field values indicated in bold. Signaling Group should be set to the value defined for the signaling group defined in the previous section. add trunk-group 15 Page 1 of 21 TRUNK GROUP Group Number: 15 Group Type: sip CDR Reports: y Group Name: AS5400 COR: 1 TN: 1 TAC: 115 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 15 Number of Members: 50 In SIP, the Session-Expires header in an INVITE indicates the desired length of the call. The SIP standard recommends that a refresher INVITE be sent at one-half the value of Session- Expires, which corresponds to the Preferred Minimum Session Refresh Interval(sec) field on Page 2. Set Preferred Minimum Session Refresh Interval(sec) to 900. Avaya Communication Manager will send 1800 for the value of Session-Expires, which will match the 1800 second default minimum session interval of the AS5400. This avoids a Session timer too small response from the AS5400 and retransmission of each call request by Avaya Communication Manager. 4 All other fields can be left at their default values. change trunk-group 15 Page 2 of 21 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): This value can also be configured to a different value on the AS5400. See Reference [5] for more details. 8 of 25

9 5.6. Define ARS Routing Use the change ars analysis command to configure route selection for outgoing calls. The entry below defines the number format to be matched and the route pattern that will be selected. For the sample configuration, dialed numbers beginning with 1 and an area code of 732 will use route pattern 15. This route pattern specifies the trunk group defined in Sections 5.4 and 5.5. change ars analysis 1 Page 1 of 2 ARS DIGIT ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd alrt n fnpa n fnpa n fnpa n 5.7. Define Route Pattern Use the change route-pattern command to define the route pattern (referenced in the previous section) that will route the PSTN call to the SIP trunk defined above. Enter the number of this trunk in Row 1 under Grp No. Enter 0 under FRL. Note that since Pfx Mrk is blank, that the prefix to the area code (e.g., 1 ) will be removed from the beginning of the dialed number. change route-pattern 15 Page 1 of 3 Pattern Number: 12 Pattern Name: 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: 15 0 n user 2: 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 M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest none 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 9 of 25

10 5.8. Define Incoming Call Treatment In the sample configuration, incoming calls from the PSTN are routed directly to a specific telephone extension based on the called number. Use the change inc-call-handling-trmt trunkgroup 15 command to specify local extensions to which incoming calls on the AS5400 SIP trunk should be terminated. Called Len and Called Number should contain the number of digits and the called number as sent by the AS5400. Del and Insert can be used to map the called number to an extension by deleting digits from the beginning and inserting the desired digits. In the sample configuration, the 5 digit incoming number (21816) is completely replaced with a new extension (34001). change inc-call-handling-trmt trunk-group 15 Page 1 of 3 INCOMING CALL HANDLING TREATMENT Service/ Called Called Del Insert Feature Len Number tie tie tie tie tie tie 5.9. Define Codec Set Use the change ip-codec-set command to define the audio codec set to be used. In the sample configuration, local calls were tested with both G.711 mu-law and G.729A. change ip-codec-set 7 Page 1 of 2 Codec Set: 7 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : 3: 4: 5: 6: 7: Media Encryption 1: none 2: 3: 10 of 25

11 On Page 2, specify the fax mode. Set FAX Mode to t.38standard or off, depending on whether the AS5400 is configured for T.38 or G.711 voice mode, respectively. See Reference [6] for more information on selecting fax modes. The value below corresponds to the T.38 fax AS5400 configuration in Section 7.1. Default values can be used for the remaining fields. change ip-codec-set 7 Page 2 of 2 IP Codec Set Allow Direct-IP Multimedia? n Mode Redundancy FAX t.38-standard 0 Modem off 0 TDD/TTY US 3 Clear-channel n 0 6. Administer Avaya SIP Enablement Services This section is only required for SES trunks to the Cisco AS5400. Avaya SES must be configured to route outbound calls to the AS5400, and inbound calls to Avaya Communication Manager. This will be accomplished by using address maps, which specify one or more dial strings to be matched and the SIP destination to which the call is to be routed if a match is found. A host address map is used to route outbound calls, and a Communication Manager Server address map is used to route inbound calls. Other standard administration functions are covered in Reference [1]. 11 of 25

12 Steps Description 1. Avaya SIP Enablement Services is configured using a web browser. Set the URL of the browser to where IP-address is the IP address of the Avaya SIP Enablement Services Edge or Edge/Home Server. On the Logon screen, enter the appropriate Logon ID and click on Logon. When prompted, enter the password. 2. The main administration screen will be displayed after logging in. Click on Launch SES Administration Interface. 12 of 25

13 Steps Description 3. Expand the Hosts link on the left side of the main administration page, and then click on List. 13 of 25

14 Steps Description 4. The existing home and edge servers will be displayed. The address map will be defined for Home 2, since Avaya Communication Manager is routing PSTN calls to Home 2. 5 In the row corresponding to Home 2 (see Figure 1), click on Map under Commands. 5. The List Host Address Map page is displayed. Click on Add Map in New Group. 5 Note that in a multi-home configuration, the address map can be defined on either Home, and Avaya Communication Manager can route calls to either Home. The Avaya SES servers will perform the appropriate routing (via the Edge) to the server containing the address map. Routing directly to the Home containing the address map is more efficient in terms of the number of messages and hops. 14 of 25

15 Steps Description 6. The Add Host Address Map page will be displayed. Specify a Name for the address map, and the Pattern match specification. In the sample configuration, 10-digit PSTN numbers with area code 732 are to be routed to the Cisco AS5400 Universal Gateway. The pattern match specification is applied to the Uniform Resource Identifier (URI) field of incoming INVITE messages. The URI usually takes the form sip:user@domain, where domain can be a domain name or an IP address. In the sample configuration, user is actually the PSTN telephone number sent by Avaya Communication Manager. An example would be sip: @companyx.com. The specification below means match on the characters sip:732 if they occur at the beginning of the URI, followed by any number of characters. This will cover PSTN numbers within the 732 area code. See Appendix A for a more detailed description of the syntax for address map patterns. Check Replace URI (default). When routing the incoming INVITE, Avaya SIP Enablement Services will replace the URI with the URI specified in the contact (see Step 8). Click on Add; then click on Continue on the confirmation page. 15 of 25

16 Steps Description 7. The current host address map list is displayed. To specify the routing for the address map just defined, click on Add Another Contact. 8. Fill in the Contact field as shown. In this case, the IP address of the AS5400 is specified, as well as the transport protocol (TCP) used to send SIP signaling messages. Avaya SIP Enablement Services will substitute $(user) with the user field (i.e., the dialed number received from Avaya Communication Manager) in the incoming INVITE message. Click on Add; then click on Continue on the confirmation page. 16 of 25

17 Steps Description 9. The List Host Address Map page is then displayed with the updated contact information. This completes configuration of the host address map for routing calls to the AS Next, inbound calls from the Cisco AS5400 must be routed to Avaya Communication Manager. A Communication Manager Server Address Map must be defined for this purpose. Expand the Communication Manager Servers link on the left side and click on List. 17 of 25

18 Steps Description 11. The Avaya Communication Manager Servers will be displayed. The address map will be defined for the server associated with Avaya SES Home 2. In the row corresponding to S8300h2 click on Map under Commands. 12. The List Communication Manager Server Address Map page is displayed. Click on Add Map in New Group. 18 of 25

19 Steps Description 13. The Add Communication Manager Server Address Map page will be displayed. Specify a Name for the address map, and the Pattern match specification. In the sample configuration, the Cisco AS5400 is sending a specific 5-digit extension, so the pattern matches on that (see Section 5.8). Check Replace URI (default). When routing the incoming INVITE, Avaya SIP Enablement Services will replace the URI with the URI specified in the contact (see Step 14). Click on Add; then click on Continue on the confirmation page. 14. The List Communication Manager Server Address Map page is displayed. Note that for Communication Manager Server address maps, the contact information is automatically populated. 19 of 25

20 Steps Description 15. Each AS5400 must be administered as a trusted host in the Avaya SES involved in call routing with that gateway. In the sample configuration, calls are routed between the AS5400 and Home 2, so a trusted host for the AS5400 must be configured there. This will permit far-end call clearing and other supplementary call service requests by AS5400 to be handled without authentication challenges. Expand the Trusted Hosts link on the left side, and select Add. Enter the IP Address of the AS5400, specify the Avaya SES Host by selecting the IP address of the Avaya SES server (Home 2 in the sample configuration), and enter an informative Comment. Click on Add; then click on Continue on the confirmation page. 20 of 25

21 Steps Description 16. The List Trusted Hosts page will be displayed, showing the newly added entry for the AS Configure Cisco AS5400 Universal Gateway The following sections provide annotated configuration steps for supporting voice/fax call routing via the Cisco AS5400 Universal Gateway. They are not a complete set of configuration steps for the product; only configuration commands relative to the sample configuration are shown. Access the AS5400 via telnet or the console interface Define Voice Services and Interfaces Note that the voice configuration commands are global and apply to all interfaces and dial-peers. voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 clid strip pi-restrict all sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 session transport tcp asymmetric payload dtmf voice class codec 1 codec preference 1 g711ulaw interface FastEthernet0/0 ip address duplex full speed auto ip route T.38 Fax (replace with fax protocol pass-through g711ulaw for voice mode) Same as above for VoIP Restricted call support Bind signaling & media to specific Ethernet port SIP signaling via TCP Allow different rcv/xmt DTMF payload types Global Codec set to g.711 mu-law (use g729r8 to specify G.729A) Network parameters for the IP interface Default route (IP gateway) 21 of 25

22 7.2. Define PSTN Interface The following command configures the T1 PRI interface to the simulated PSTN. isdn switch-type primary-ni voice call send-alert controller T1 2/0 framing esf linecode b8zs pri-group timeslots 1-24 voice-port 2/0:D interface Serial2/0:23 no ip address encapsulation hdlc isdn timer T isdn timer T isdn incoming-voice modem isdn T309-enable no cdp enable Set ISDN PRI switch protocol Send ALERT message (rather than PROGRESS) back on receipt of SETUP message. Configure T1 controller 0, module slot 2 Use all 23 bearer channels Switch to voice-port config mode using PRI D Configure PRI signaling channel (last channel) DISCONNECT timer CALL PROCEEDING timer Enable call clearing timer No Cisco Discovery Protocol 7.3. Define Call Routing The following commands specify call routing to Avaya SES Home 2. Note that the AS5400 will match a dialed number to the destination-pattern specified in the dial peers to determine which dial peer to use for routing the call. In the sample configuration, the simulated PSTN sends the last 5 digits of the called number (e.g., 21816) for incoming calls from the PSTN. For SES SIP trunks, the session target ipv4: command causes incoming calls to be routed to the SES Home 2. For direct trunk, this would be replaced with session target ipv4: to cause routing to Avaya Communication Manager. dial-peer voice voip service session destination-pattern progress_ind setup enable 1 session protocol sipv2 session target ipv4: dtmf-relay rtp-nte voice-class codec 1 dial-peer voice 732 pots service session destination-pattern direct-inward-dial port 2/0:D forward-digits 10 prefix *9 VoIP dial peer Inbound PSTN calls come into the AS5400 with this extension. SIP signaling Route to Avaya SES Home 2 Use RFC 2833 for signaling DTMF Codec preferences (See Section 5.9) T1/PRI dial peer Outbound calls for area code 732 T1 signaling interface Forward AAA-NNN-XXXX Prefix *9 for outside calls 22 of 25

23 8. Verification Steps All features shown in Table 3 were tested using both SES and direct SIP trunks in the sample configuration. The following steps can be used to verify/troubleshoot installations in the field. 1. Place a call from a PSTN phone to an extension on Avaya Communication Manager/Avaya SES. Use the list trace tac command to confirm that the call arrives on the configured SIP trunk in Avaya Communication Manager. Use a call that permits verification of DTMF support (e.g., let the call to go to voice mail). 2. Place a call from an Avaya Communication Manager extension to a PSTN phone. Use the list trace tac command to confirm that the call is routed outbound on the configured SIP trunk in Avaya Communication Manager. Use a call that permits verification of DTMF support (e.g., let the call to go to voice mail). 3. If calls fail in an SES trunk configuration, check to make sure that address maps are properly configured in Avaya SES. Use a network sniffer to verify that INVITE messages are properly routed between the Avaya SES and Cisco AS5400. If DTMF is not successfully transmitted, check proper configuration of the Cisco AS5400 (see Section 7.1 and Reference [7]). 4. Connect fax machines to a PSTN analog port and to an analog port on Avaya Media Gateways, as applicable. Confirm successful calls and accurate fax transmissions. If they fail, use the previous verification steps to verify that the calls are properly routed, and that T.38 fax mode is achieved, if applicable. Note that G.711 voice mode fax may not work in networks experiencing excessive packet loss and jitter. 9. Support For technical support of Cisco products: Internet: tac@cisco.com Telephone: Additional References [1] Installing, Administering, Maintaining, and Troubleshooting SIP Enablement Services, Issue 5, Doc ID , January, 2008, available at [2] Avaya Extension to Cellular and OPS Installation and Administration Guide, Version 6.0 Issue 9, Doc ID , June 2005, available at [3] SIP Support in Avaya Communication Manager Running on Avaya S83xx Servers, Issue 8, Doc ID , January, 2008, available at [4] Administrator Guide for Avaya Communication Manager, Issue 4, Doc ID , January 2008, available at [5] Cisco AS5350 and Cisco AS5400 Universal Gateway Software Configuration Guide, available at [6] Cisco IOS Fax Services over IP Roadmap, available at [7] Configuring SIP DTMF Features, available at 23 of 25

24 Appendix A Address Map Pattern Strings in Avaya SIP Enablement Services The syntax for the pattern is a Linux regular expression that will match the extensions to be mapped. Regular expressions are a textual way to describe pattern matching. The regular expression is a string containing a combination of normal text characters, which match themselves, and special metacharacters, which may represent items like quantity, location or types of character(s). For example, [0-9] represents any single digit and * represents any number of digits or characters. For example, ^sip:538[0-9]* would match any SIP INVITE message (^ matches the beginning of a line) for any extension 3 or more digits in length, beginning with the digits 538, and ending with any other sequence of digits. Square brackets contain a selection of characters to be matched, with a hyphen indicating a range. In the example, [0-9] matches any digit, or as another example, [13579] matches odd-numbered digits. Braces containing a whole number match that number of instances of the preceding item. For example, [0-9]{4} matches any four digits. Note that the braces may require preceding \ escape characters: \{4\}. Another helpful metacharacter is dot (period), which matches any single character. For example, the regular expression.* matches any quantity (including zero) of any character(s). 24 of 25

25 Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at 25 of 25