Transcript
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 interoperate with Avaya SIP Enablement Services and Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 Analog VoIP Gateway provides voice technology to connect analog telephones, fax machines and modems to 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 MP124 Analog VoIP Gateway at the customer enterprise sites to register with Avaya SIP Enablement Services at the service provider data center, effectively making the analog endpoints appear as SIP endpoints. These analog endpoints were then able to utilize the telephony features of Avaya Communication Manager. Information in these Application Notes has been obtained through DeveloperConnection compliance testing and additional technical discussions. Testing was conducted via the DeveloperConnection Program at the Avaya Solution and Interoperability Test Lab.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
1 of 45 AudioCodesMP124
1. Introduction These Application Notes describe the configuration procedures required for the AudioCodes MP-124 Analog VoIP Gateway to interoperate with Avaya SIP Enablement Services (SES) and Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 Analog VoIP Gateway provides voice technology to connect analog telephones, fax machines and modems to 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 Analog VoIP Gateway at the customer enterprise sites to register with Avaya SIP Enablement Services server at the service provider data center, effectively making the analog endpoints appear as SIP endpoints. These analog endpoints were then able to utilize the telephony features of Avaya Communication Manager. The AudioCodes MP-124 Analog VoIP Gateway (MP-124) incorporates 24 analog ports for connection to telephones, fax machines and modems, supporting 24 simultaneous VoIP calls.
1.1. 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 SES server, an Avaya G650 Media Gateway, and an Avaya S8500 Media Server running Avaya Communication Manager and Avaya IA770 Intuity Audix. The access to the PSTN for all enterprise customers was provided by an ISDN PRI trunk from the Avaya G650 Media Gateway. Each enterprise site consists of a MP-124 with the analog ports connected to analog telephones, and a fax machine or a modem. The MP-124 is connected to the service provider data center network over an IP network via an Avaya C363T-PWR Converged Stackable Switch. Avaya 4600 Series IP Telephones (H.323) were also located at the enterprise sites to verify calls to and from the analog endpoints behind the MP-124. These Avaya 4600 Series IP Telephones registered directly with Avaya Communication Manager.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
2 of 45 AudioCodesMP124
Figure 1: Reference Test Configuration
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
3 of 45 AudioCodesMP124
2. Equipment and Software Validated The following equipment and software/firmware were used for the test configuration provided. Equipment Avaya S8500 Media Server including Avaya IA770 Intuity Audix
Avaya G650 Media Gateway Avaya SIP Enablement Services Avaya 4600 Series IP Telephones (H.323) Avaya 363T-PWR Converged Stackable Switch AudioCodes MP-124 Analog VoIP Gateway Analog telephones Fax machine PCs with modem
Software/Firmware Avaya Communication Manager 3.1.2 (R013x.01.2.632.1) To support T.38 fax, the following post–GA release was used: (R013x.01.2.635.0) 25.23.0 3.1 (3.1.0.0-018.0) 2.3 (4620SW) 4.5.14 4.80A.018.007 Windows 2000 Professional
3. Configure Avaya Communication Manager This section describes the necessary configuration on Avaya Communication Manager to interoperate with the SIP interface of 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.
3.1. 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 MP-124 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 and then to the MP-124. As a result, the SIP trunk group configured in this section carries all the SIP signaling bound to the service provider data center from the MP-124 at the customer enterprise sites and vice versa. The following steps were performed at the service provider data center in the configuration shown in Figure 1.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
4 of 45 AudioCodesMP124
Step 1.
Description 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 MP-124 will use two SIP trunks. A call between an analog telephone behind the 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 OPTIONAL FEATURES
Page
IP PORT CAPACITIES Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP eCons: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable H.323 Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: Maximum Number of DS1 Boards with Echo Cancellation: Maximum TN2501 VAL Boards: Maximum G250/G350/G700 VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports:
800 2400 0 0 0 0 0 0 800
USED 0 4 0 0 0 0 0 0 200
0 1 0 0 0 0
0 0 0 0 0 0
2 of
10
(NOTE: You must logoff & login to effect the permission changes.)
2.
Use the change node-names ip command to assign the node name and IP address for Avaya SES at the service provider data center. In this case, ses and 192.168.10.50 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 192.168.10.43 are the name and IP address assigned to the CLAN IP interface on the Avaya G650 Media Gateway. change node-names ip Name CLAN MEDPRO default procr ses
CTM; Reviewed: SPOC 4/19/2007
Page IP NODE NAMES IP Address Name 192.168.10 .43 192.168.10 .42 0 .0 .0 .0 192.168.10 .40 192.168.10 .50
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
1 of
1
IP Address . . . . . . . . . . . . . . .
5 of 45 AudioCodesMP124
Step 3.
Description 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 calls. 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 MP-124 as well as any IP telephone 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 (not shown) 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: 65531 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
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
6 of 45 AudioCodesMP124
Step 4.
Description 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 to the 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 include at least one codec configured on the MP-124. The example below shows the values used in the tested configuration. change ip-codec-set 1
Page
1 of
2
IP Codec Set Codec Set: 1 Audio Codec 1: G.711MU 2: G.729AB 3: 4:
Silence Suppression n n
Frames Per Pkt 2 2
Packet Size(ms) 20 20
If T.38 fax is desired, on Page 2 enter t.38-standard in the FAX Mode field. For data modem calls, retain the default value of off for the Modem Mode field. change ip-codec-set 1
Page
2 of
2
IP Codec Set Allow Direct-IP Multimedia? n
FAX Modem TDD/TTY Clear-channel
CTM; Reviewed: SPOC 4/19/2007
Mode t.38-standard off US n
Redundancy 0 0 3 0
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
7 of 45 AudioCodesMP124
Step 5.
Description 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 configurations that 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 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 2833. The default values for the other fields may be used. add signaling-group 1
Page
1 of
1
SIGNALING GROUP Group Number: 1
Group Type: sip Transport Method: tls
Near-end Node Name: CLAN Near-end Listen Port: 5061
Far-end Node Name: ses 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
CTM; Reviewed: SPOC 4/19/2007
Direct IP-IP Audio Connections? y IP Audio Hairpinning? y
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
8 of 45 AudioCodesMP124
Step 6.
Description 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 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: Group Name: Direction: Dial Access? Queue Length: Service Type:
1 siproute two-way n 0 tie
Group Type: sip COR: 0 Outgoing Display? n
TN: 1
CDR Reports: y TAC: 701
Night Service: Auth Code? n Signaling Group: 1 Number of Members: 100
7.
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 TRUNK FEATURES ACA Assignment? n
Page
3 of
21
Measured: none Maintenance Tests? y
Numbering Format: public Prepend '+' to Calling Number? n
Replace Unavailable Numbers? n
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
9 of 45 AudioCodesMP124
Step 8.
Description Use the change route-pattern command to configure a route pattern that routes to the 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 trunk group number configured in Step 6. 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 Pattern Number: 1 Pattern Name: siproute SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted No Mrk Lmt List Del Digits Dgts 1: 1 0 2: 3: 4: 5: 6:
9.
Page
1 of
3
DCS/ QSIG Intw n n n n n n
IXC
user user user user user user
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 routes to the 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 No. 1: Main 2: 3: 4:
CTM; Reviewed: SPOC 4/19/2007
Timezone Rule Offset + 00:00 0 : : :
NPA
ARS FAC
Attd FAC
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
Prefix
Proxy Sel. Rte. Pat. 1
10 of 45 AudioCodesMP124
3.2. Configure SIP Stations The analog endpoints (e.g., analog telephones, fax machines and modems) behind the MP-124 are considered SIP stations from the point of view of Avaya Communication Manager. This section describes the steps for configuring SIP stations. SIP stations are administered as off-PBX station (OPS) stations. Many advanced PBX features are available to OPS stations such as call park, call pickup, and priority calls. These features are activated by dialing feature access codes (FAC) or by dialing special extensions known as feature name extensions (FNE). For more details, see [6] and [7]. The configuration for feature access codes and feature name extensions is not in the scope of this document. Step 1.
Description 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 OPTIONAL FEATURES G3 Version: V13 Location: 1 Platform: 12
1 of
10
RFA System ID (SID): 1 RFA Module ID (MID): 1
Platform Maximum Ports: Maximum Stations: Maximum XMOBILE Stations: Maximum Off-PBX Telephones - EC500: Maximum Off-PBX Telephones OPS: Maximum Off-PBX Telephones - SCCAN:
CTM; Reviewed: SPOC 4/19/2007
Page
3200 2400 0 2400 2400 0
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
USED 231 21 0 0 16 0
11 of 45 AudioCodesMP124
Step 2.
Description Use the add station x command, where x is an unused extension, to create a station extension as follows: Leave the default value the 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 alpha-numeric name. The default values may be retained for all other fields. add station 2346241
Page
1 of
4
STATION Extension: Type: Port: Name:
2346241 6408D+ X CM CUST A Analog 2
STATION OPTIONS Loss Group: Data Module? Speakerphone: Display Language:
2 n 2-way english
Lock Messages? n Security Code: Coverage Path 1: Coverage Path 2: Hunt-to Station:
BCC: TN: COR: COS:
0 1 1 1
Personalized Ringing Pattern: 1 Message Lamp Ext: 2346241 Mute Button Enabled? y
Media Complex Ext: IP SoftPhone? n
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
12 of 45 AudioCodesMP124
Step 3.
Description On Page 3, assign the buttons under BUTTON ASSIGNMENTS as follows: Make sure that at least two call-appr buttons are administered. Assign a no-hld-cnf button to enable the station to activate the Conference on Answer feature. Assign an 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 2346241
Page
3 of
4
STATION SITE DATA Room: Jack: Cable: Floor: Building:
Headset? Speaker? Mounting: Cord Length: Set Color:
ABBREVIATED DIALING List1: group
1
BUTTON ASSIGNMENTS 1: call-appr 2: call-appr 3: 4: no-hld-cnf
4.
List2:
n n d 0
List3:
5: auto-cback 6: 7: 8:
Use the add off-pbx-telephone station-mapping command to associate an administered extension with an external phone number as follows: Set the Station Extension field to the extension administered in Step 2. Set the Application field to OPS. In the Phone Number field, enter the phone number of the off-PBX telephone. The off-PBX telephone is the analog telephone behind the MP-124. Refer to Section 5 Steps 11-12. 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 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Extension 2346241
CTM; Reviewed: SPOC 4/19/2007
Application OPS
Dial Phone Number Prefix - 2346241 -
Trunk Selection 1
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
Page
1 of
2
Configuration Set 1
13 of 45 AudioCodesMP124
Step 5.
Description 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. Leave other fields to their default values. change off-pbx-telephone configuration-set 1
Page
1 of
1
CONFIGURATION SET: 1 Configuration Set Description: Calling Number Style: CDR for Origination: CDR for Calls to EC500 Destination? Fast Connect on Origination? Post Connect Dialing Options: Cellular Voice Mail Detection: Barge-in Tone? Calling Number Verification? Identity When Bridging:
6.
1 network phone-number y y dtmf none n y principal
Repeat Steps 1-4 for each analog endpoint behind the MP-124.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
14 of 45 AudioCodesMP124
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.
Description Access the Avaya SES administration web interface, by entering http://
/admin as the URL in an Internet browser, where 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.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
15 of 45 AudioCodesMP124
Step 2.
Description The Avaya SES Administration main page shown below will be displayed.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
16 of 45 AudioCodesMP124
Step 3.
Description 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.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
17 of 45 AudioCodesMP124
Step 4.
Description 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 the same as the Avaya SES server. After configuring the Edit System Properties page, click the Update button (not shown).
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
18 of 45 AudioCodesMP124
Step 5.
Description 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 (not shown). In the Host IP Address field, enter the IP address of Avaya SES. In the DB Password field, enter the 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 at the bottom of the page (not shown).
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
19 of 45 AudioCodesMP124
Step 6.
Description From the left pane of the administration web interface, expand the Media Servers 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 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.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
20 of 45 AudioCodesMP124
Step 7.
Description Inbound SIP calls arriving at Avaya SES are routed to the appropriate Avaya Communication Manager for termination services. In the compliance test configuration, which has a single Avaya Media Server, calls originating from SIP endpoints registered to the SES (like the MP-124) will be automatically routed to the Avaya Media Server. Thus, strictly speaking, the Media Server Address Maps described below are not necessary. However, these maps were present during the testing and thus are described below for completeness. Media Server Address Maps route calls by comparing the Uniform Resource Identifier (URI) of an incoming INVITE message to the pattern configured in the Media Server Address Map. 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. The next step shows an example of an address map to match the dialing of an Avaya Communication Manager FAC. In the compliance test, 3-digit numbers beginning with digit 8 were configured as FACs.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
21 of 45 AudioCodesMP124
Step 7. cont’d
Description 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 (not shown). Click on the Map link associated with the appropriate media server to display the List Media Server Address Map page (not shown). 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 FACs ^sip:8[09]{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. Appendix A provides a detailed description of the syntax for address map patterns. Click the Add button.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
22 of 45 AudioCodesMP124
Step 8.
Description After submitting the previous form, the List Media Server Address Map page appears as follows, showing an entry for the newly created Address Map. Notice that the list includes additional Address Maps. These were created to support the use of other commonly used dial strings (Abbreviated Dial Codes, etc.). The procedures used to create these additional Address Maps are similar to those shown in Step 7, but are not included here. In addition, a Media Server Contact is created automatically after the first Media Server Address Map is added. The contact specifies the IP address of the C-LAN board (in this example, 192.168.10.43) and the transport protocol used to send SIP signaling messages. The user in the original request URI is substituted for $(user).
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
23 of 45 AudioCodesMP124
Step 9.
Description From the left pane of the administration web interface, expand the Users option and select Add to add a user. Specify a value for the 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 the same as the User Name configured on the MP-124 for an analog telephone. Refer to Section 5, Step 12. Specify a value for the Password field. Enter the same value in the Confirm Password field. 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 MP-124 for an analog telephone. See Section 5, Step 12. The Host field is populated with the 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 to Step 11. Click the Add button.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
24 of 45 AudioCodesMP124
Step 10.
Description 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 MP-124.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
25 of 45 AudioCodesMP124
Step 13.
Description 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.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
26 of 45 AudioCodesMP124
5. Configure AudioCodes MP-124 Analog VoIP Gateway The MP-124 is configured via an Internet browser using the embedded web server interface. Both of the MP-124 gateways shown in Figure 1 are configured using the procedures in this section. The following steps describe how to configure: Step 1.
IP address of the MP-124 and associated 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 Description Access the MP-124 embedded web server interface, by entering http:/// as the URL in an Internet browser, where is the default IP address of the MP-124. Refer to [8] for the default IP address.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
27 of 45 AudioCodesMP124
Step 2.
Description 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 MP-124, under IP Configuration enter values for the IP Address, Subnet Mask, and Default Gateway IP Address fields. To assign basic SIP parameters, under SIP Parameters assign the parameters as shown below: Set the Working with Proxy field to Yes to enable the interworking of the MP124 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 a REGISTER request to Avaya SES according to the Authentication Mode parameter. Refer to Step 5 for Authentication Mode. Click the Reset button and click OK in the prompt. The MP-124 applies the changes and restarts. After the MP-124 restarts, access the MP-124 embedded web server interface, by entering http:/// as the URL in an Internet browser, where is the new IP address of the MP-124 configured in this step.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
28 of 45 AudioCodesMP124
Step 3.
4.
Description From the left pane of the web interface, click the Protocol Management option. The right pane displays configuration options as shown below.
SIP Proxy & Registration From the right pane of the web interface, click the Protocol Definition Æ Proxy & Registration option as shown below.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
29 of 45 AudioCodesMP124
Step 5.
Description The basic SIP proxy and registration parameters were configured in Step 2. Scroll down the SIP Proxy & Registration page to configure additional SIP parameters as shown below. Set the Registration Time to 3600 to specify that the registration for the analog endpoints behind the 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 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.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
30 of 45 AudioCodesMP124
Step 6.
Description General Parameters From the right pane of the web interface, click the Protocol Definition Æ General Parameters option. Configure general SIP parameters as shown below. Select the Enable Early Media field as Enable. If enabled, the MP-124 sends a 183 Session Progress response with Session Description Protocol (SDP) 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 15 for other T.38 fax settings.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
31 of 45 AudioCodesMP124
Step 7.
Description Coders (Codecs) From the right pane of the web interface, click the Protocol Definition Æ Coders option. From the Coders screen, configure up to five preferred coders (and their attributes) for the MP-124. 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 the 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 codec 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 codec setting in Avaya Communication Manager. The default values may be retained for all other fields. After completing this screen, click the Submit button.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
32 of 45 AudioCodesMP124
Step 8.
Description DTMF and Dialing From the right pane of the web interface, click the Protocol Definition Æ 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 1st 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 MP-124 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.
CTM; Reviewed: SPOC 4/19/2007
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 FACs in Avaya Communication Manager and 3xx for FNEs in Avaya Communication Manager. The default values may be retained for all other fields. After completing this screen, click the Submit button.
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
33 of 45 AudioCodesMP124
Step 9.
Description Advanced Control Parameters From the right pane of the web interface, click the Advanced Parameters Æ 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. 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).
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
34 of 45 AudioCodesMP124
Step 10.
Description Supplementary Services From the right pane of the web interface, click the Advanced Parameters Æ Supplementary Services option. Enable services such as 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 MP-124. 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.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
35 of 45 AudioCodesMP124
Step 11.
Description 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 MP-124, to which the analog endpoints are connected. Channel is a port number on the back of the 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 2346241 is assigned to channel 1. Refer to Section 3.2 and Section 4, Steps 9-14 for how to configure related telephone extensions in Avaya Communication Manager and Avaya SES. The default values may be retained for all other fields. After completing this screen, click the Submit button (not shown).
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
36 of 45 AudioCodesMP124
Step 12.
Description Endpoint Authentication From the right pane of the web interface, click the Endpoint Settings Æ Authentication option. The Authentication table defines a user name and password combination for authentication for each MP-124 port as shown below. Enter a value for the User Name and Password fields 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).
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
37 of 45 AudioCodesMP124
Step 13.
Description Enable Caller ID From the right pane of the web interface, click the Endpoint Settings Æ 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 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 10). 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).
14.
From the left pane of the web interface, click the Advanced Configuration option. The right pane displays configuration options as shown below.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
38 of 45 AudioCodesMP124
Step 15.
Description T.38 Fax From the right pane of the web interface, click the Media Settings Æ 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 T.38 fax is desired. Use the default values for all other fields. After completing this screen, click the Submit button. Refer to Step 6 for other T.38 fax settings.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
39 of 45 AudioCodesMP124
Step 16.
17.
Description For proper modem interoperability, an additional parameter must be changed in the ini file. From a web browser, enter /AdminPage where is the IP address of the MP-124. The main page will appear as shown below.
Select ini Parameters in the left pane. In the Parameter Name field, select ModemBypassPayloadType from the pull-down menu. Enter 0 in the Enter Value field. Click Apply New Value. A value of 0 in the payload type indicates a G.711MU call. By default, the MP-124 uses a proprietary value for the payload type of modem calls so these calls can be distinguished from other G.711MU calls.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
40 of 45 AudioCodesMP124
6. Interoperability Compliance Testing This section describes the compliance testing used to verify the interoperability between the AudioCodes MP-124 Analog VoIP Gateway, Avaya SIP Enablement Services and Avaya Communication Manager. This section covers the general test approach and the test results.
6.1. General Test Approach The general test approach was to make calls to/from the telephones connected to the MP-124 at the enterprise sites using various codec settings and exercising common PBX features. The calls were made to/from the PSTN and within the enterprise site.
6.2. Test Results The MP-124 successfully passed compliance testing. The following features and functionality were verified. Calls to/from the PSTN Calls between the MP-124 analog telephones Calls between the MP-124 analog telephones and the Avaya 4600 Series IP Telephones (H.323) G.711MU and G.729AB codec support Proper recognition of DTMF transmissions MP-124 support for hold, transfer (attended and unattended), and call waiting Proper operation of voicemail with message waiting indicators (MWI); for the analog phones, MWI was provided via stutter dial tone Call forwarding provided by Avaya Communication Manager Conferencing (Avaya H.323 IP telephone initiates a conference that includes an MP-124 analog endpoint) A subset of Avaya Communication Manager features (e.g., hunt group, call pickup group, call forward, and send all calls) involving the analog telephones behind the MP-124 T.38 fax support Modem calls Proper system recovery after a MP-124 restart The following observations were made during the compliance test: To support T.38 fax with the MP-124, a post-GA release (R013x.01.2.635.0) of Avaya Communication Manager was required. The MP-124 does not support initiating a conference using flash hook. In addition, the Avaya Communication Manager “conference on answer” feature did not work for this combined Avaya/AudioCodes solution. This feature was tested with the GA release.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
41 of 45 AudioCodesMP124
7. 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. Verify that an analog telephone behind the 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 MP-124. Verify that a call can be placed between an analog telephone behind the MP-124 and a telephone in the PSTN. Verify that a call can be placed between an analog telephone behind the MP-124 and an Avaya 4600 Series IP Telephone (H.323). Verify that a T.38 fax can be transmitted between a fax machine behind the MP-124 and a fax machine in the PSTN. Verify that a data modem call can be placed between a data modem behind the MP-124 and a data modem in the PSTN. Verify that the transfer and hold features can be successfully activated from the analog telephones behind the 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 pickup group, call forward, and send all calls) that involve the analog telephones behind the MP-124.
8. Support Technical support for the AudioCodes MP-124 Analog VoIP Gateway can be obtained from AudioCodes. See the Support link at www.audiocodes.com for contact information.
9. Conclusion The analog telephones, modems and fax machines behind the AudioCodes MP-124 Analog VoIP Gateway can successfully register to Avaya SIP Enablement Services, place/receive calls and utilize telephony features of Avaya Communication Manager.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
42 of 45 AudioCodesMP124
10. Additional References The following Avaya product documentation can be found at http://support.avaya.com. [1] Feature Description and Implementation For Avaya Communication Manager, Issue 4.0, February 2006, Document Number 555-245-205. [2] Administrator Guide for Avaya Communication Manager, Issue 2.1, May 2006, Document Number 03-300509. [3] Installing and Administering SIP Enablement Services R3.1, Issue 1.5, February 2006 Document Number 03-600768. [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 555-245-206. [5] 4600 Series IP Telephone Release 2.4 LAN Administrator Guide, April 2006, Issue 2.3, Document Number 555-233-507. [6] Avaya Extension to Cellular User’s Guide, February 2006, Issue 9, Document Number 210-100-700. [7] Avaya Extension to Cellular and OPS Installation and Administration Guide, June 2005, Issue 9, Document Number 210-100-500. The following MP-124 product documentation is available from AudioCodes. Visit http://www.audiocodes.com for company and product information. [8] AudioCodes MediaPack MP-124 and MP-11x User’s Manual 4.8.
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
43 of 45 AudioCodesMP124
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 caret 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 1+ 10 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:[email protected]:5060;transport=udp SIP/2.0
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
44 of 45 AudioCodesMP124
©2007 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 e-mail 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 DeveloperConnection Program at [email protected].
CTM; Reviewed: SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes ©2007 Avaya Inc. All rights reserved.
45 of 45 AudioCodesMP124
