Application Notes for AudioCodes MP-124 Analog VoIP Gateway with Avaya SIP Enablement Services and Avaya Communication Manager Issue 1.

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for AudioCodes MP-124 Analog VoIP Gateway with Avaya SIP Enablement Services and Avaya Communication Manager Issue 1.0 Abstract These Application Notes describe the configuration procedures required for the AudioCodes MP-124 Analog VoIP gateway to successfully interoperate with Avaya SIP Enablement Services and Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 gateway provides voice technology to connect analog telephones, fax machines and modems to the SIP servers and IP based PBX systems. In the reference test network simulating a hosted telephony solution, this technology enabled the analog endpoints behind the AudioCodes MP- 124 at the customer enterprise sites to register with Avaya SIP Enablement Services server at the service provider data center, effectively making them SIP endpoints. These analog endpoints were then able to utilize the telephony features of Avaya Communication Manager. 1 of 44

2 1. Introduction These Application Notes describe the configuration procedures required for the AudioCodes MP-124 Analog VoIP gateway to successfully interoperate with Avaya SIP Enablement Services and Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 gateway provides voice technology to connect analog telephones, fax machines and modems to the SIP servers and IP based PBX systems. In the reference test network simulating a hosted telephony solution, this technology enabled the audio endpoints behind the AudioCodes MP-124 at the customer enterprise sites to register with Avaya SIP Enablement Services server at the service provider data center, effectively making them SIP endpoints. These analog endpoints were then able to utilize the telephony features of Avaya Communication Manager. The MP-124 gateway incorporates up to 24 analog ports for connection to telephones, fax machines and modems, supporting up to 24 simultaneous VoIP calls Reference Test Network Figure 1 shows the reference configuration used for simulating a hosted telephony solution with two customer enterprise sites and a service provider data center. The service provider data center site consists of an Avaya SIP Enablement Services (SES) server, an Avaya S8500 Media Server running Avaya Communication Manager and an Avaya G650 Media Gateway. The access to the PSTN for all the enterprise customers was provided by an ISDN PRI trunk from an Avaya G650 Media Gateway to the PSTN. Each enterprise site consists of an AudioCodes MP-124 analog VoIP gateway, with the analog ports connected to the analog telephones, a fax machine and a modem. The AudioCodes MP-124 is connected to the service provider data center network over an IP network via an Avaya C363T switch. Avaya 4600 Series H.323 IP telephones were also located at the enterprise sites to verify calls to and from the analog endpoints behind MP-124 gateway. Avaya 4600 Series H.323 IP telephones registered directly with Avaya Communication Manager. 2 of 44

3 Figure 1: Reference Test Configuration 3 of 44

4 2. Equipment and Software Validated The following equipment and software/firmware were used for the test configuration provided. Equipment Software/Firmware Avaya S8500 Media Server Avaya Communication Manager (R013x ) Avaya G650Media Gateway (Media Gateway Processor) Avaya SES 3.1 ( ) Avaya 4600 Series IP (H.323) Telephones 2.3 (4620SW) Analog Telephones - Avaya 363T Switch AudioCodes MP-124 analog VoIP gateway 4.80A Fax machine - PCs with modem Windows Configure Avaya Communication Manager This section describes the necessary configuration on Avaya Communication Manager to interoperate with the SIP interface of the Avaya SES. The following configuration of Avaya Communication Manager was performed using the System Access Terminal (SAT). After completion of the configuration in this section, perform a save translation command to make the changes permanent Configure SIP Trunking This section describes the steps for configuring a SIP trunk group between Avaya Communication Manager and Avaya SES. Avaya SES acts as a SIP proxy for the service provider data center in Figure 1. Thus, the AudioCodes MP-124 analog SIP gateway will direct all SIP traffic bound for the service provider data center to the Avaya SES located there. Avaya SES will then forward the inbound SIP traffic to Avaya Communication Manager. The path is the same in the reverse direction with SIP signaling flowing from Avaya Communication Manager to Avaya SES to the AudioCodes MP-124. As a result, the SIP trunk group configured in this section carries all the SIP signaling and associated media streams bound to the service provider data center from the AudioCodes MP-124 gateways at the customer enterprise sites and vice versa. These steps were performed at the service provider data center in the configuration shown in Figure 1. 4 of 44

5 1. Use the display system-parameters customer-options command to verify that sufficient SIP trunk capacity exists. On Page 2, verify that the number of SIP trunks supported by the system is sufficient for the number of SIP trunks needed. Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. Thus, a call between two analog telephones behind the AudioCodes MP-124 VoIP gateway will use two SIP trunks. A call between an analog telephone behind the AudioCodes MP-124 and a non-sip service provider (e.g., PSTN trunk) will only use one SIP trunk. The license file installed on the system controls the maximum permitted. If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales representative to make the appropriate changes. 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: 0 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: 1 0 Maximum G250/G350/G700 VAL Sources: 0 0 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 44

6 2. Use the change node-name ip command to assign the node name and IP address for Avaya SES at the service provider data center. In this case, ses and are being used, respectively. The node name ses will be used throughout the other configuration forms of Avaya Communication Manager. In this example, CLAN and are the name and IP address assigned to the CLAN IP interface on Avaya G650 Media Gateway. change node-names ip Page 1 of 1 IP NODE NAMES Name IP Address Name IP Address CLAN MEDPRO default procr ses of 44

7 3. Use the change ip-network-region n command, where n is the number of the region to be changed, to define the connectivity settings for all VoIP resources and IP endpoints within the IP region. Select an IP network region that will contain the Avaya SES server. The association between this IP Network Region and the Avaya SES server will be done on the Signaling Group form as shown in Step 5. In the case of the tested configuration, the same IP network region that contains the Avaya S8500 Media Server was selected to contain the Avaya SES server. By default, the Media Server is in IP Network Region 1. On the IP Network Region form: The Authoritative Domain field is configured to match the domain name configured on Avaya SES. In this configuration, the domain name is test.com. This name will appear in the From header of SIP messages originating from this IP region. By default, IP-IP Direct Audio (shuffling) is enabled to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya G650 Media Gateway. This is true for both intra-region and inter-region IP-IP Direct Audio. Shuffling can be further restricted at the trunk level on the Signaling Group form. The Codec Set is set to the number of the IP codec set to be used for calls within this IP network region. In this configuration, this codec set will apply to all calls to and from the AudioCodes MP-124 as well as any IP phone (H.323 or SIP) within the enterprise. If different IP network regions are used for the Avaya S8500 Media Server and the Avaya SES server, then Page 3 of each IP Network Region form must be used to specify the codec set for inter-region communications. The default values can be used for all other fields. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: Authoritative Domain: test.com Name: CM 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: 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: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 7 of 44

8 4. Use the change ip-codec-set n command, where n is the codec set value specified in Step 3, to enter the supported audio codecs for calls routed to Avaya SES and the AudioCodes MP-124. Multiple codecs can be listed in priority order to allow the codec to be negotiated during call establishment. The list should include the codecs the enterprise wishes to support within the normal trade-off of bandwidth versus voice quality. It must also include the preferred codec configured on the AudioCodes MP The example below shows the values used in the tested configuration. change ip-codec-set 1 Page 1 of 2 Codec Set: 1 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : G.729AB n : 4: Page forward to Page 2. If the T.38 faxing is desired, enter t.38-standard in the FAX field. For data modem calls, retain the default value of off for the Modem field. change ip-codec-set 1 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 8 of 44

9 5. Use the add signaling group n command, where n is the number of an unused signaling group, to create the SIP signaling group as follows: Set the Group Type field to sip. The Transport Method field will default to tls (Transport Layer Security). TLS is the only link protocol that is supported for communication between Avaya SES and Avaya Communication Manager. Specify the node name CLAN on Avaya S8500 Media Server and the node name ses for Avaya SES Server as the two ends of the signaling group in the Near-end Node Name and the Far-end Node Name fields, respectively. These field values are taken from the IP Node Names form shown in Step 2. In alternative configurations, which do not use a CLAN board, the near (local) end of the SIP signaling group will be the procr instead of the CLAN board. Ensure that the recommended TLS port value of 5061 is configured in the Near-end Listen Port and the Far-end Listen Port fields. In the Far-end Network Region field, enter the IP Network Region value assigned in the IP Network Region form in Step 3. This defines which IP network region contains the Avaya SES server. If the Far-end Network Region field is different from the near-end network region, the preferred codec will be selected from the IP codec set assigned for the inter-region connectivity for the pair of network regions. Enter the domain name of Avaya SES in the Far-end Domain field. In this configuration, the domain name is test.com as configured in Step 3. This domain is specified in the Uniform Resource Identifier (URI) of the SIP To header in the INVITE message. The Direct IP-IP Audio Connections field is set to y to enable the direct audio path (shuffling) between the two IP endpoints. The DTMF over IP field must be set to the default value of rtp-payload for a SIP trunk. This value enables Avaya Communication Manager to send DTMF transmissions using RFC The default values for the other fields may be used. change signaling-group 1 Page 1 of 1 SIGNALING GROUP Group Number: 1 Group Type: sip Transport Method: tls Near-end Node Name: CLAN Far-end Node Name: ses Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: 1 Far-end Domain: test.com Bypass If IP Threshold Exceeded? y DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y IP Audio Hairpinning? y 9 of 44

10 6. Add a SIP trunk group by using the add trunk-group n command, where n is the number of an unused trunk group. For the compliance test, trunk group number 1 was chosen. On Page 1, set the fields to the following values: Set the Group Type field to sip. Choose a descriptive Group Name. Specify an available trunk access code (TAC) that is consistent with the existing dial plan. Set the Service Type field to tie. Specify the signaling group associated with this trunk group in the Signaling Group field as previously specified in Step 5. Specify the Number of Members supported by this SIP trunk group. As mentioned earlier, each SIP call between two SIP destinations (whether internal or external) requires two SIP trunks for the duration of the call. Thus, a call between two SIP extensions, e.g., two telephones behind the AudioCodes MP- 124, will use two SIP trunks. A call between a SIP extension and the PSTN will only use one SIP trunk. The default values may be retained for the other fields. add trunk-group 1 Page 1 of 21 TRUNK GROUP Group Number: 1 Group Type: sip CDR Reports: y Group Name: siproute COR: 0 TN: 1 TAC: 701 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 1 Number of Members: On Page 3: Set the Numbering Format field to public. This field specifies the format of the calling party number sent to the far-end. The default values may be retained for the other fields. add trunk-group 1 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 10 of 44

11 8. Configure a route pattern that is routing to its outbound SIP proxy server (Avaya SES server). This will be assigned on the locations screen. Refer to Step 9. The example below shows the route pattern used in the reference test configuration. The Pattern Name can be any descriptive name. The Grp No. is set to the trunkgroup number for the trunk to be used. The FRL field defines the facility restriction level for this route pattern. The value of 0 is the least restrictive. Default values for all other fields can be used. change route-pattern 1 Page 1 of 3 Pattern Number: 1 Pattern Name: siproute 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: 1 0 n user 2: n user 3: n user 4: n user 5: n user 6: n user 9. Use the change locations command to assign the route pattern to the location. This screen allows for each location to point to the route pattern that is routing to its outbound SIP proxy server. This correlation is required by features and services such as Transfer and URI Dialing. Only one location created by default, known as Main, exists for the service provider data center and the customer sites. Enter the route pattern number from the previous step in the Proxy Sel. Rte Pat. field. The default values may be retained for all other fields. change locations Page 1 of 16 LOCATIONS ARS Prefix 1 Required For 10-Digit NANP Calls? y Loc. Name Timezone Rule NPA ARS Attd Pre- Proxy Sel. No. Offset FAC FAC fix Rte. Pat. 1: Main + 00: : 3: 4: : 11 of 44

12 3.2. Configure SIP Stations The analog endpoints (e.g. analog telephones, fax machines and modems) behind the AudioCodes MP-124 are considered SIP stations from the point of view of Avaya Communication Manager. This section describes the steps for configuring SIP stations. Support for SIP stations that use SIP trunks allows any fully compliant SIP telephone to interoperate with Avaya telephones. This means that any SIP telephone can: Dial and be dialed as an extension in the enterprise dial plan. Put calls on hold and participate in transfers and conference calls. SIP stations that are administered as Outboard Proxy SIP stations, also known as off-pbx station (OPS) stations, support most Extended Access Features, such as call park, call pick-up, and priority calls. To activate these features, either dial Feature Access Codes or dial special extensions, that is, Feature Name Extensions. For more details, see Avaya Extension to Cellular User s Guide, Issue 6, DocID , and the Avaya Extension to Cellular and OPS Installation and Administration Guide, Issue 8, DocID The configuration for Feature Access Codes and Feature Name Extensions is not in the scope of this document. These steps were performed using the SAT at the service provider data center in the configuration shown in Figure 1 to add the OPS stations. 12 of 44

13 1. Use the display system-parameters customer-options command to verify that Maximum Off-PBX Telephones OPS has been set to a value that will accommodate the number of telephones to be supported. display system-parameters customer-options Page 1 of 10 OPTIONAL FEATURES G3 Version: V13 Location: 1 RFA System ID (SID): 1 Platform: 12 RFA Module ID (MID): 1 USED Platform Maximum Ports: Maximum Stations: Maximum XMOBILE Stations: 0 0 Maximum Off-PBX Telephones - EC500: Maximum Off-PBX Telephones - OPS: Maximum Off-PBX Telephones - SCCAN: 0 0 (NOTE: You must logoff & login to effect the permission changes.) 2. Use the add station x command, where x is an unused extension to create a station extension as follows: Leave the default value for Type field to 6408D+. Set the Port field to X. The value X implies that this station is being administered without hardware such as a port on an Avaya analog or a digital circuit pack. Set the Name field to any alphna-numeric name. add station Page 1 of 4 STATION Extension: Lock Messages? n BCC: 0 Type: 6408D+ Security Code: TN: 1 Port: X Coverage Path 1: COR: 1 Name: CM CUST A Analog 2 Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Loss Group: 2 Personalized Ringing Pattern: 1 Data Module? n Message Lamp Ext: Speakerphone: 2-way Mute Button Enabled? y Display Language: english Media Complex Ext: IP SoftPhone? n 13 of 44

14 3. Page forward to Page 3 and assign the buttons under BUTTON ASSIGNMENTS as follows: Make sure that at least two call-appr buttons are administered. Assign no-hld-cnf button to enable the station to activate the Conference on Answer feature. Assign auto-cback button to enable the station to activate the Automatic CallBack feature. Note that these are virtual button assignments that are used internally by Avaya Communication Manager to implement the associated features. add station Page 3 of 4 STATION SITE DATA Room: Headset? n Jack: Speaker? n Cable: Mounting: d Floor: Cord Length: 0 Building: Set Color: ABBREVIATED DIALING List1: group 1 List2: List3: BUTTON ASSIGNMENTS 1: call-appr 5: auto-cback 2: call-appr 6: 3: 7: 4: no-hld-cnf 8: 14 of 44

15 4. Use the add off-pbx-telephone station-mapping to associate an administered extension with an external phone number as follows: Set the Station Extension field to the extension administered in Step 2. The Application field indicates the type of off-pbx application associated with an internal phone extension. Set this field to OPS to indicate SIP Enablement Services application. In the Phone Number field, enter the phone number of the off-pbx telephone. The off-pbx telephone is the analog telephone behind the AudioCodes MP-124. Refer to Section 5 Steps Set the Trunk Selection field to 1, same as the trunk group configured in Section 3.1, Step 6. The Configuration Set field indicates the configuration set number that defines the desired call treatment options. Set this to 1. See Step 5. add off-pbx-telephone station-mapping Page 1 of 2 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Application Dial Phone Number Trunk Configuration Extension Prefix Selection Set OPS Use the change off-pbx-telephone configuration-set n command, where n is a configuration set number to assign the call treatment options as follows: Set the Fast Connect on Origination field to y to allow Avaya Communication Manager to do some additional processing prior to connecting the call. Leave other fields to their default values. change off-pbx-telephone configuration-set 1 Page 1 of 1 CONFIGURATION SET: 1 Configuration Set : 1 Calling Number Style: network CDR for Origination: phone-number CDR for Calls to EC500 Destination? y Fast Connect on Origination? y Post Connect Dialing Options: dtmf Cellular Voice Mail Detection: none Barge-in Tone? n Calling Number Verification? y Identity When Bridging: principal 6. Repeat Steps 1-4 for each analog endpoint behind the AudioCodes MP of 44

16 4. Configure Avaya SES This section covers the configuration of Avaya SES. Avaya SES is configured via an Internet browser using the administration web interface. It is assumed that Avaya SES software and the license file have already been installed on the server. During the software installation, the installation script is run from the Linux shell of the server to specify the IP network properties of the server along with other parameters. For additional information on these installation tasks, refer to [3]. Step 1. Access the Avaya SES administration web interface, by entering as the URL in an Internet browser, where <ip-addr> is the IP address of Avaya SES. Log in with the appropriate credentials and then select the Launch Administration Web Interface link from the main page as shown below. 16 of 44

17 2. The Avaya SES Administration Home Page shown below will be displayed. 17 of 44

18 3. After making changes within Avaya SES, it is necessary to commit the database changes using the Update link that appears when changes are pending. Perform this step by clicking on the Update link found in the bottom of the blue navigation bar on the left side of any of the Avaya SES administration pages as shown below. It is recommended that this be done after making each set of changes described in the following steps. 18 of 44

19 4. From the left pane of the administration web interface, expand the Server Configuration option and select System Properties. This page displays the software version in the SES version field and the network properties entered via the installation script during the installation process. On the Edit System Properties page: Enter the SIP Domain name assigned to Avaya SES. This must match the Authoritative Domain field configured on Avaya Communication Manager shown in Section 3.1 Steps 3 and 5. Enter the License Host field. This is the host name, the fully qualified domain name, or the IP address of the SIP proxy server that is running the WebLM application and has the associated license file installed. In this case, the WebLM server is same as the Avaya SES server. After configuring the Edit System Properties page, click the Update button (not shown). 19 of 44

20 5. After setting up the domain on the Edit System Properties page, create a host computer entry for Avaya SES. The following example shows the Edit Host page since the host had already been added to the system. The Edit Host page shown below is accessible by clicking on the Hosts link in the left pane and then clicking on the edit option under the Commands section of the subsequent page that is displayed. Enter the Logical IP or Logical Name of this server in the Host IP Address, the IP address of Avaya SES. Enter the DB Password that was specified while running the installation script during the system installation. Verify the Host Type field. Since only one Avaya SES server exists in the test configuration, the Avaya SES server provides the functionality of both a home and edge server. Thus, the Host Type is configured as home/edge. The default values for the other fields may be used. Click the Update button. 20 of 44

21 6. From the left pane of the administration web interface, expand the Media Server option and select Add to add the Avaya Media Server to the list of media servers known to Avaya SES. Adding the media server will create the Avaya SES side of the SIP trunk previously created in Avaya Communication Manager. On the Add Media Server Interface page, enter the following information: A descriptive name in the Media Server Interface Name field (e.g., s8500clan). In the Host field, select the IP address of the Avaya SES server from the pulldown menu that will serve as the SIP proxy for this media server. Since there is only one Avaya SES server in this configuration, the Host field is set to the host shown in Step 5. Select TLS (Transport Link Security) for the Link Type. TLS provides encryption at the transport layer. TLS is the only link protocol that is supported for communication between Avaya SES and Avaya Communication Manager. Enter the IP address of the C-LAN in the SIP Trunk IP Address field. In alternative configurations, that do not use a C-LAN board, the SIP Trunk IP Address would be the IP address of the Avaya Media Server. The default values may be retained for all other fields. After completing the Add Media Server Interface page, click the Add button. 21 of 44

22 7. Inbound SIP calls arriving at Avaya SES are routed to the appropriate Avaya Communication Manager for termination services. This routing is specified in a Media Server Address Map configured on Avaya SES. This routing compares the Uniform Resource Identifier (URI) of an incoming INVITE message to the pattern configured in the Media Server Address Map, and if there is a match, the call is routed to the designated Avaya Communication Manager. The URI usually takes the form of sip:user@domain, where domain can be a domain name or an IP address. Patterns must be specific enough to uniquely route incoming calls to the proper destination if there are multiple Avaya Media Servers supported by the Avaya SES server. In this test configuration, some of the incoming calls from customer sites require a Media Server Address Map entry. For example, in the test configuration, 3-digit numbers beginning with digit 8 were configured as the Avaya Communication Manager Feature Access Codes. From customer sites, the user portion of the SIP URI will contain the Feature Access Code. An example of a SIP URI in an INVITE message received from a customer site would be: sip: 841@ In this example, the user portion is the Feature Access Code (FAC) 841. Thus, the media server address map strategy was to define pattern matches for 8xx in the URI, where x is any digit between 0 and 9, and have Avaya SES forward the messages that match to the appropriate media server. To configure a Media Server Address Map: Expand the Media Servers option in the left pane of the administration web interface and select List. This will display the List Media Servers page. Click on the Map link associated with the appropriate media server to display the List Media Server Address Map page. Click on the Add Map In New Group link. The page shown below is displayed. The Host field displays the name of the media server to which this map applies. Enter a descriptive name in the Name field. Enter the regular expression to be used for the pattern matching in the Pattern field. The example below shows the pattern specification for Feature Access Codes ^sip:8[0-9]{2}. This expression will match any SIP URI that begins with the text string sip:8 followed by a 2-digit string. Based on the value of the Host field, these SIP calls will then be routed to host S8500clan. 22 of 44

23 Appendix A provides a detailed description of the syntax for address map patterns. Click the Add button once the form is completed. 23 of 44

24 8. After configuring the media server address map, the List Media Server Address Map page appears as shown below, showing an entry for 2xx Address Map. In addition to the dialed phone numbers, the Media Server Address Map may also be needed for other user dialable digit strings such as Abbreviated Dial number or a Feature Access Code administered on Avaya Communication Manager. The list shows other such maps (e.g., AbbDialing for abbreviated dialing) the patterns are not shown. After the first Media Server Address Map is added, the Media Server Contact is created automatically. For the Media Server Address Map that was added, the following contact was created and displayed in the Contact field: sip:$(user)@ :5061;transport=tls The contact specifies the IP address of the CLAN board on Avaya S8500 Media Server and the transport protocol used to send SIP signaling messages. The user in the original request URI is substituted for $(user). 24 of 44

25 9. From the left pane of the administration web interface, expand the User option and select Add to add a user and an extension for an analog endpoint behind the AudioCodes MP-124. Specify a Primary Handle and User ID. A handle specifies a user on Avaya SES. The User ID is used to authenticate a user on Avaya SES. The User ID must be same as the User ID configured on the AudioCodes MP-124 for an analog telephone. Refer Section 5, Step 11. Specify a Password and Confirm Password. The Password along with the User ID is used to authenticate a user on Avaya SES. The Password must be same as the Password configured on the AudioCodes MP-124 for an analog telephone. See Section 5, Step 11. The Host field is pre-populated with that Avaya SES Host IP Address since there is only one Avaya SES in the test configuration. Enter any descriptive First Name and Last Name. Select Add Media Server Extension to add an OPS extension for the user. Refer Step 11. Click the Add button. 25 of 44

26 10. The following Continue page appears, confirming that the user was added successfully. Click Continue. 11. The Add Media Server Extension page will appear as shown below. Enter the Extension, same as the OPS extension configured in Avaya Communication Manager. Refer to Section 3.2, Step 2. Select the appropriate Media Server configured in Step 6. Click Add. 12. Repeat Steps 9-11 to add the users and their associated extensions for additional analog endpoints behind the AudioCodes MP of 44

27 13. To view the configured users, from the left pane of the administration web interface, expand the Users option and select List. The List Users page will appear as shown below. 14. To view the configured extensions, from the left pane of the administration web interface, expand the Media Server Extensions option and select List. The List Media Server Extensions page will appear as shown below. 27 of 44

28 5. Configure AudioCodes MP-124 VoIP Gateway AudioCodes-MP124 is configured via an Internet browser using the Embedded Web Server interface. The following steps describe how to configure: IP address of the AudioCodes MP-124 and associating SIP proxy (Avaya SES). SIP proxy (Avaya SES) & registration. General Parameters Coders DTMF and Dialing Advanced Control Parameters Supplementary Services such as Hold and Transfer. Endpoint Phone Numbers. Endpoint Authentication. Enable Caller ID T.38 Fax Step 1. Access the AudioCodes MP-124 Embedded Web Server interface, by entering as the URL in an Internet browser, where <ip-addr> is the default IP address of the AudioCodes MP-124. Refer to [6] for the default IP address. 28 of 44

29 2. IP address and SIP Proxy (Avaya SES) The web interface screen appears as shown below with Quick Setup selected. To assign an IP address for the AudioCodes-MP124, under IP Configuration enter the IP Address, Subnet Mask, and Default Gateway IP Address To assign basic SIP parameters, under SIP Parameters assign the parameters as shown below: Set Working with Proxy field to Yes to enable the interworking of the AudioCodes MP-124 with Avaya SES proxy server. Enter the IP address of the Avaya SES in the Gateway Name, Proxy IP Address and Proxy Name fields. Select Enable Registration to Enable. The AudioCodes will send REGISTER request to Avaya SES according to the Authentication Mode parameter. Refer to Step 4 for Authentication Mode. Click Reset button and click OK in the prompt. The AudioCodes MP-124 applies the changes and restarts. After the AudioCodes MP-124 restarts, access the AudioCodes MP-124 Embedded Web Server interface, by entering as the URL in an Internet browser, where <ip-addr> is the new IP address of the AudioCodes MP-124 configured in this step. 29 of 44

30 3. From the left pane of the web interface, click the Protocol Management option. The right pane displays configuration options as shown below. 4. SIP Proxy & Registration From the right pane of the web interface, click the Protocol Definition Proxy & Registration option as shown below. 30 of 44

31 The basic SIP proxy and registration parameters were configured in Step 2. Configure additional SIP parameters as shown below. Set the Registration Time to 3600 to specify that the registration for the analog endpoints behind the AudioCodes MP-124 is valid for one hour. Select Send All Invite to Proxy field to Yes so that all the Invite requests including those generated as a result of transfer or redirect are sent to Avaya SES. Enter any alphanumeric string in the Cnonce field. This field is used for the mutual authentication of the server and client. Select the Authentication Mode as Per Endpoint. This implies that the registration and authentication of each audio endpoint behind the AudioCodes MP-124 with Avaya SES is done separately. The default values may be retained for all other fields. After completing this screen, click the Submit button. 31 of 44

32 5. General Parameters From the right pane of the web interface, click the Protocol Definition General Parameters option. Configure general SIP parameters as shown below. Select the Enable Early Media field as Enable. If enabled, the AudioCodes MP-124 gateway sends 183 Session Progress response with Session Protocol parameters(instead of 180 Ringing), allowing the media stream to be set up prior to the answering of the call. Select Fax Signaling Method field as T.38 Relay if the T.38 fax transmission is desired. Select the Use user=phone in SIP URL field as No. The default values may be retained for all other fields. After completing this screen, click the Submit button (not shown here). Refer to Step 14 for other T.38 fax settings. 32 of 44

33 6. Coders From the right pane of the web interface, click the Protocol Definition Coders option. From the Coders screen, configure the first to fifth preferred coders (and their attributes) for the AudioCodes MP-124 gateway. The first coder is the highest priority coder and is used by the gateway whenever possible. If the far end gateway (on Avaya Communication Manager) cannot use the coder assigned as the first coder, the gateway attempts to use the next coder and so forth. Configure Coder parameters as shown below. From the Coder Name drop-down list, select the appropriate coder. Refer to Section 3.1, Step 4 for the coder selection and preferred order in Avaya Communication Manager. For G.729 coder, enable the silence suppression by selecting Enable from the Silence Suppression drop down list. With silence suppression enabled, the gateway includes the string annexb=yes in the SDP of the relevant SIP messages, an equivalent of G.729AB. Refer to Section 3.1, Step 4 for the coder setting in Avaya Communication Manager. The default values may be retained for all other fields. After completing this screen, click the Submit button. 33 of 44

34 7. DTMF and Dialing From the right pane of the web interface, click the Protocol Definition DTMF & Dialing option. Use this screen to configure parameters that are associated with DTMF and dialing. Configure the DTMF and Dialing parameters as shown below. Enter the maximum number of digits that can be dialed (e.g., 18) in the Max Digits in Phone Num field. From the 1 st Tx DTMF Option drop-down list, select RFC 2833 as the preferred transmit DTMF negotiation method. For the RFCC 2833 Payload Type field, enter 101. Assign the digit map pattern in the Digit Mapping Rules field. If the digit string (dialed number) has matched one of the patterns in the digit map, the AudioCodes MP-124 gateway stops collecting digits and starts to establish a call with the collected number. The digit map pattern contains up to 52 options separated by a vertical bar ( ). The maximum length of the entire digit pattern is limited to 152 characters. For example: 2xxxxxx 1xx 91xxxxxxxxxx 8xx 3xx was used in the test configuration, where 2xxxxxx was used for dialing telephone extensions, 1xx was used for speed dialing (Abbreviated Dialing feature in Avaya Communication Manager), 91xxxxxxxxxx for PSTN calls, 8xx for Feature Access Codes in Avaya Communication Manager and 3xx for Feature Name Extensions in Avaya Communication Manager. The default values may be retained for all other fields. After completing this screen, click the Submit button. 34 of 44

35 8. Advanced Control Parameters From the right pane of the web interface, click the Advanced Parameters General Parameters option. Configure the advanced control parameters as shown below: From the Enable Current Disconnect drop down list (under Disconnect and Answer Supervision), select Enable. The AudioCodes MP-124 gateway generates a Current Disconnect Pulse to an analog endpoint behind the AudioCodes MP-124 gateway when the call on this endpoint is released from the Avaya SES. Set Broken Connection Timeout to desired amount of time an RTP packet isn t received, after which the call is disconnected. Set the Max Number of Active Calls to the maximum number of channels available on the gateway. For the tested configuration, it was set to 24. The default values may be retained for all other fields. After completing this screen, click the Submit button (not shown here). 35 of 44

36 9. Supplementary Services From the right pane of the web interface, click the Advanced Parameters Supplementary Services option. Enable the services like hold and transfer as shown below: From the Enable Caller ID drop down list select Enable. When enabled, the calling number and name are sent to the analog endpoints behind the AudioCodes MP-124 gateway. The default values may be retained for all other fields. Hold, Transfer, Call Forwarding and Call Waiting are enabled by default. After completing this screen, click the Submit button. 36 of 44

37 10. Endpoint Phone Numbers From the right pane of the web interface, click the Endpoint Phone Numbers option. Enable and assign telephone numbers for the VoIP gateway ports on the AudioCodes MP-124, to which the analog endpoints are connected. Channel is a port number on the back of the AudioCodes MP-124. To enable a channel, enter a port number. In the Phone Number field, enter the telephone number assigned to that channel. For example, the telephone number is assigned to channel 1. Refer to Section 3.2 and Section 4 Step 9-14 for how to configure related telephone extensions in Avaya Communication Manager and Avaya SES. After completing this screen, click the Submit button (not shown here). 37 of 44

38 11. Endpoint Authentication From the right pane of the web interface, click the Endpoint Settings Authentication option. The Authentication Table defines a username and password combination for authentication for each AudioCodes MP-124 port as shown below: Enter the User Name and Password for each port. This is used for registration and authentication with Avaya SES. Make sure that these values are the same as configured using Section 4, Step 9. After completing this screen, click the Submit button (not shown here). 38 of 44

39 12. Enable Caller ID From the right pane of the web interface, click the Endpoint Settings Generate Caller ID to Tel option. The Generate Caller ID to Tel table is used to enable or disable (per port) the Caller ID generation and detection as shown below. From the Caller ID drop down list for each port, select Enable or Disable. In the tested configuration the Caller ID was enabled for the Ports 1-3. When enabled, an analog telephone with a caller ID display capability displays the calling number and name on receiving calls beginning with the second ring cycle. The Caller ID type defaults to BellCore - see Step 9. If set to other call types, the caller ID display and behavior may be different. The default values may be retained for all other fields. After completing this screen, click the Submit button (not shown here). 13. From the left pane of the web interface, click the Advanced Configuration option. The right pane displays configuration options as shown below. 39 of 44

40 14. T.38 Fax From the right pane of the web interface, click the Media Settings Fax/Modem/CID Settings option and configure the Fax option as shown below: From the Fax Transport Mode drop down list, select T.38 Relay if the T.38 faxing is desired. Use the default values for all other fields. After completing this screen, click the Submit button. Refer to Step 5 for other T.38 fax settings. 6. Verification Steps This section provides verification steps that may be performed to verify that the solution described in these Application Notes is configured properly. Verify that the SIP trunk group is in-service. To do this, use the status trunk n command, where n is the number of the trunk group to be verified. Verify that the SIP signaling group is in-service. To do this, use the status signalinggroup n command, where n is the number of the signaling group to be verified. 40 of 44

41 Verify that an analog telephone behind the AudioCodes MP-124 can register with Avaya SES. To do this, from the left pane of the administration web interface of Avaya SES, expand the User option and select Registered Users. Verify that a call can be placed between two analog telephones behind the AudioCodes MP-124. Verify that a call can be placed between an analog telephone behind the AudioCodes MP- 124 and a telephone in the PSTN. Verify that a call can be placed between an analog telephone behind the AudioCodes MP- 124 and an Avaya 4600 Series H.323 IP telephone. Verify that a fax can be transmitted between a fax machine behind the AudioCodes MP- 124 and a fax machine in the PSTN. Verify that a data modem call can be placed between a data modem behind the AudioCodes MP-124 and a data modem in the PSTN. Verify that a T.38 fax can be transmitted between a fax machine behind the AudioCodes MP-124 and a fax machine in the PSTN. Verify that the conference, transfer, and hold features can be successfully activated from the analog telephones behind the AudioCodes MP-124. Verify that the DTMF digits can be transmitted over an established call. Verify a sample of other Avaya Communication Manager features (e.g., hunt group, call pick up group, call forward, send all calls, and call waiting) that involve the analog telephones behind the AudioCodes MP-124. The following exceptions were observed during the testing. The analog telephones behind the AudioCodes MP-124 cannot activate conference. For T.38 Fax, an unofficial patch was required on Avaya Communication Manager to make it work. The fix will be available in the next Generally Available release of Avaya Communication Manager. The following Avaya Communication Manager features worked, however the audible alerting was not exactly as expected: o Call forward unconditional: the calls are forwarded but no alerting on the forwarded (analog) station. o Send All Calls (SAC): SAC d station hears one full ring instead of an abbreviated ring or tone. 7. Support Technical support for the AudioCodes MP-124 can be obtained from AudioCodes. See the Support link at for contact information. 8. Conclusion The analog telephones, modems and fax machines behind the AudioCodes MP-124 can successfully register to Avaya SIP Enablement Services (SES) and utilize telephony features of Avaya Communication Manager. 41 of 44

42 9. Additional References The following Avaya product documentation can be found at [1] Feature and Implementation For Avaya Communication Manager, Issue 4.0, February 2006, Document Number [2] Administrator Guide for Avaya Communication Manager, Issue 2.1, May 2006, Document Number [3] Installing and Administering SIP Enablement Services R3.1, Issue 1.5, February 2006 Document Number [4] SIP Support in Release 3.1 of Avaya Communication Manager Running on the Avaya S8300, S8500, S8500B, S8700, and S8710 Media Server, February 2006, Issue 6, Document Number [5] 4600 Series IP Telephone Release 2.4 LAN Administrator Guide, April 2006, Issue 2.3, Document Number The following AudioCodes MP-124 product documentation is available from AudioCodes. Visit for company and product information. [6] AudioCoded MediaPack MP-124 and MP-11x User s Manual of 44

43 APPENDIX A: Specifying Pattern Strings in Address Maps The syntax for the pattern matching used within Avaya SES is a Linux regular expression used to match against the URI string found in the SIP INVITE message. Regular expressions are a way to describe text through 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). In the pattern matching string used in Avaya SES: Normal text characters and numbers match themselves. Common metacharacters used are: o A period. matches any character once (and only once). o An asterisk * matches zero or more of the preceding characters. o Square brackets enclose a list of any character to the matched. Ranges are designated by using a hyphen. Thus, the expression [12345] or [1-5] both describe a pattern that will match any single digit between 1 and 5. o Curley brackets containing an integer n indicate that the preceding character must be matched exactly n time. Thus, 5{3} matches 555 and [0-9]{10} indicates any 10 digit number. o The circumflex character ^ as the first character in the pattern indicates that the string must begin with the character following the circumflex. Putting these constructs together as used in this document, the pattern to match the SIP INVITE string for any valid digit number in the North American dial plan would be: ^sip:1[0-9]{10} This reads as: Strings that begin with exactly sip:1 and having any 10 digits following will match. A typical INVITE request below uses the shaded portion to illustrate the matching pattern. INVITE sip: @ :5060;transport=udp SIP/ of 44

44 2006 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please any questions or comments pertaining to these Application Notes along with the full title and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at 44 of 44