Asm Acm As5400

Avaya Solution & Interoperability Test Lab
Configuring SIP IP Telephony Using Avaya AuraTM Session Manager, Avaya AuraTM Communication Manager, and the Cisco AS5400 Universal Gateway Issue 1.0
Abstract
These Application Notes describe the configuration steps required to connect the Cisco AS5400 Universal Gateway to a SIP infrastructure consisting of an Avaya AuraTM Session Manager and an Avaya S8500 Server with G6500 Media Gateway running Avaya AuraTM Communication Manager. SIP trunking with Communication Manager and AS5400 are connected via Session Manager. The configuration steps described are also applicable to other Linux-based Avaya Servers and Media Gateways running Avaya AuraTM Communication Manager.
AL; Reviewed: SPOC 10/6/2009
Solution & Interoperability Test Lab Application Notes 2009 Avaya Inc. All Rights Reserved.
1 of 28 ASM-ACM-AS5400
1. Introduction
With the introduction of the SIP standard that supports telephony as well as other communication modes, there is a much broader range of SIP telephones and gateways available to customers. There will be sales opportunities involving customers who wish to purchase the Avaya SIP offer, but have a significant investment in SIP/PSTN gateways other than those offered by Avaya. Customers may be interested in replacing their existing telephony infrastructure with Avaya servers, but wish to re-use the existing nonAvaya gateways. These Application Notes describe the configuration steps for using the Cisco AS5400 Universal Gateway with Avaya Aura Session Manager, and Avaya Aura Communication Manager. Only the configuration steps pertinent to interoperability of Cisco and Avaya equipment are covered. General administration information can be found in the product documentation as well as the specific references listed in Section 9. The configuration described should be applicable to other Linux-based Avaya Servers and Media Gateways running Avaya Aura Communication Manager.
2. Configuration
The configuration used as an example in these Application Notes is shown in Figure 1. The diagram indicates logical signaling connections. With the exception of the Avaya 6408D Digital Telephone and the fax machines, all components are physically connected to an Ethernet Switch and are administered on a single subnet. The Cisco AS5400 Universal Gateway is connected to the PSTN network using a T1 PRI. Note: There is a DEFINITY PBX connected between the PSTN and the Cisco AS5400 that removes the first 5 digit from an incoming PSTN call before presenting a 5-digit extension to the Cisco AS500. This DEFINITY PBX is not shown to in the figure, and does not affect the interoperability of the overall solution. The configuration steps described in the following sections cover the routing of calls by Session Manager, Communication Manager, and the AS5400. The AS5400 is configured as a peer gateway that routes calls between its T1/PRI and SIP interfaces. The SIP trunk to the Cisco AS5400 is shown between the Cisco AS5400 and Avaya Session Manager.
Figure 1: Sample Network Configuration
3. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration: DEVICE DESCRIPTION Avaya Aura Communication Manager - Running on an Avaya S8300 Server with an Avaya G430 Media Gateway Avaya Aura System Manager - Running on an Avaya S8510 Server Avaya Aura Session Manager - Running on an Avaya S8510 Server Avaya 9630 IP Telephone (H.323) Avaya 6408D Digital Telephone Cisco AS5400 Universal Gateway Cisco 7965 Unified IP Phone (SIP) Cisco 7912 Unified IP Phone (SCCP) VERSION TESTED R 5.2 (R015x.02.0.947.3) SP1 (02.0.947.3-17294) 1.0 1.1 (1.1.4.0.111013) 3.0 12.4(15)T7 SIP45.8-4-1S App Load ID CP7912080003SCCP070409A Boot Load ID LD0100BOOT021112A
4. Supported Features
Table 1 gives a summary of the features supported by the configuration steps described in these Application Notes. . Feature Extension-to-extension calls Basic calls to legacy phones Caller ID DTMF Fax (G.711 voice mode) Fax (T.38) Call Hold, Resume Attended Transfer Unattended Transfer Conference Compressed codecs Comments Digital, H.323, Analog RFC 2833
G.729
Table 1: SIP Telephony Feature Support
5. Configure Avaya Aura Communication Manager

This section shows the configuration of Avaya Aura Communication Manager. All configurations in this section are administered using the System Access Terminal (SAT). These Application Notes assumed that the basic configuration has already been administered. For further information on Avaya Aura Communication Manager, please consult with references [4] and [5]. 1. Use the display system-parameter customer options command to compare the Maximum Administered SIP Trunks field value with the corresponding value in the USED column. The difference between the two values needs to be greater than or equal to the desired number of simultaneous SIP trunk connections. Note: The license file installed on the system controls the maximum features permitted. If there is insufficient capacity or a required feature is not enabled, contact an authorized Avaya sales representative to make the appropriate changes.
display system-parameters customer-options OPTIONAL FEATURES 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 Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: 50 450 450 450 0 0 100 100 100 Page 2 of 10
USED 6 1 0 0 0 0 0 0 20
2. Use the change system-parameters features command to allow for trunk-to-trunk transfers. This feature is needed to allow for transferring an incoming/outgoing call from/to a remote switch back out to the same or different switch. For simplicity, the Trunk-toTrunk Transfer field was set to all to enable all trunk-to-trunk transfers on a system wide basis. Note: This feature poses significant security risk, and must be used with caution. As an alternative, the trunk-to-trunk feature can be implemented using Class Of Restriction or Class Of Service levels.
change system-parameters features Page 1 of FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 10 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? y Music/Tone on Hold: music Type: ext 14383 Music (or Silence) on Transferred Trunk Calls? no DID/Tie/ISDN/SIP Intercept Treatment: attd 18
3. Use the change node-names ip command to add entries for the CLAN and Avaya Aura Session Manager that will be used for connectivity. In the sample network, CLAN and 172.28.10.7 are entered as Name and IP Address for the Avaya Aura Communication Manager system. In addition, ASM-SM1 and 10.1.2.170 are entered for Avaya Aura Session Manager.
change node-names ip IP NODE NAMES Name ASM-SM1 ASM-SM2 CLAN IP Address 10.1.2.170 10.1.2.180 172.28.10.7 Page 1 of 2
4. Use the change ip-network-region n command, where n is the network region number to configure the network region being used. In the sample network ipnetwork-region 1 is used. For the Authoritative Domain field, enter the SIP domain name configured for this enterprise and a descriptive Name for this ip-networkregion. Set Intra-region IP-IP Direct Audio and Inter-region IP-IP Direct Audio to yes to allow for direct media between endpoints. Set the Codec-Set to 1 to use ipcodec-set 1.
change ip-network-region 1 IP NETWORK REGION Region: 1 Location: Authoritative Domain: avaya.com Name: ASM-to-Cisco 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: 3329 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 Page 1 of 19
5. Use the change ip-codec-set n command, where n is the existing codec set number to configure the desired audio codec. One page 2 of this form, set Fax Mode to t.38-standard.
change ip-codec-set 1 IP Codec Set Codec Set: 1 Audio Codec 1: G.711MU 2: G.729 Silence Suppression n n Frames Per Pkt 2 2 Packet Size(ms) 20 20 Page 1 of 2
change ip-codec-set 1 IP Codec Set Allow Direct-IP Multimedia? n
Page
2 of
FAX Modem TDD/TTY Clear-channel
Mode t.38-standard off US n
Redundancy 0 0 3 0
6. In the test configuration, signal group 25 along with trunk group 25 were used to reach Session Manager. Use the add signaling-group n command, where n is the signaling-group number being added to the system. The Far-end Domain is left blank so that the signaling group accepts any authoritative domain.
display signaling-group 25 SIGNALING GROUP Group Number: 25 IMS Enabled? n Group Type: sip Transport Method: tls
Near-end Node Name: CLAN Near-end Listen Port: 5061 Far-end Domain: DTMF over IP: rtp-payload Session Establishment Timer(min): 3 Enable Layer 3 Test? n H.323 Station Outgoing Direct Media? n
Far-end Node Name: ASM-SM1 Far-end Listen Port: 5061 Far-end Network Region: 1 Direct IP-IP Audio Connections? IP Audio Hairpinning? Direct IP-IP Early Media? Alternate Route Timer(sec): y n n 6
7. Use the add trunk-group n command, where n is the new trunk group number being added to the system. The following screens show the settings used for trunk group 25.
Add trunk-group 25 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 25 To-ASM two-way n 0 tie Group Type: sip CDR Reports: y COR: 1 TN: 1 TAC: 125 Outgoing Display? n Night Service: Auth Code? n Signaling Group: 25 Number of Members: 10 add trunk-group 25 ACA Assignment? n TRUNK FEATURES Measured: none Maintenance Tests? y Numbering Format: public UUI Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Show ANSWERED BY on Display? Y Page 3 of 21 Page 1 of 21
8. Use the change public-unknown-numbering command to define the calling party number to be sent out to through the SIP trunk. Add an entry for the trunk group defined in Step 7. In the sample network configuration below, all calls originating from a 5-digit extension beginning with 11 and routed to trunk group 25 will result in a 5-digit calling number. The calling party number will be in the SIP From header.
change public-unknown-numbering 0 Page 1 of NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: 3 5 11 25 5 Maximum Entries: 240 2
9. Use the change inc-call-handling-trmt trunk-group n where n is the incoming trunk group number to map the incoming digit to an internal extension.
change inc-call-handling-trmt trunk-group 25 INCOMING CALL HANDLING TREATMENT Service/ Number Number Del Insert Feature Len Digits tie 5 21816 5 11010 Page 1 of 30
10. Configure a FEATURE ACCESS CODE to access ARS.

display feature-access-codes Page 1 of FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: Abbreviated Dialing List2 Access Code: Abbreviated Dialing List3 Access Code: Abbreviated Dial - Prgm Group List Access Code: Announcement Access Code: *10 Answer Back Access Code: *16 Attendant Access Code: Auto Alternate Routing (AAR) Access Code: 8 Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2: Automatic Callback Activation: *11 Deactivation: #10 Call Forwarding Activation Busy/DA: *20 All: *21 Deactivation: #20 Call Forwarding Enhanced Status: Act: Deactivation: 6
11. Use the change ars analysis n command where n is the dial string pattern to configure an ars entry for Dialed String 1732 to use Route Pattern 25.
change ars analysis 1732 ARS DIGIT ANALYSIS TABLE Location: all Dialed String 1732 174 175 Total Min Max 11 11 11 11 11 11 Route Pattern 25 deny deny Call Type fnpa fnpa fnpa Node Num Page 1 of 2 1
Percent Full: ANI Reqd n n n
12. Configure a route pattern to correspond to the newly added SIP trunk group. Use the change route-pattern n command, where n is the route pattern number specified in Step 11. Configure this route pattern to route calls to trunk group number 25 configured in Step 7.
change route-pattern 25 Pattern Number: 25 Pattern Name: To-ASM SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted No Mrk Lmt List Del Digits Dgts 1: 25 0 2: 3: BCC VALUE TSC CA-TSC 0 1 2 M 4 W Request 1: y y y y y n 2: y y y y y n n n ITC BCIE Service/Feature PARM rest rest Page 1 of 3
DCS/ QSIG Intw n n n
IXC user user user
No. Numbering LAR Dgts Format Subaddress none none
13. Use the save translation command to save all changes.

save translation SAVE TRANSLATION Command Completion Status Success Error Code 0
6. Configuring Avaya Aura Session Manager

This section provides the procedures for configuring Avaya Aura Session Manager. It is assumed that the basic configuration for Avaya Aura Session Manager has already been completed. For further information on Avaya Aura Session Manager, please consult with references [1], [2], and [3] . The following areas are configured in this section. SIP domain Adaptations SIP Entities corresponding to the SIP telephony systems and Avaya Aura Session Manager Entity Links, which define the SIP trunk parameters used by Avaya Aura Session Manager when routing calls to/from SIP Entities Time Ranges during which routing policies are active Routing Policies, which control call routing between the SIP Entities Dial Patterns, which govern how a call is routed Avaya Aura Session Manager, corresponding to the Avaya Aura Session Manager Server to be managed by Avaya Aura System Manager
1. Access the Avaya Aura System Manager using a Web Browser and entering http://<ip-address>/IMSM, where <ip-address> is the IP address of Avaya Aura System Manager. Log in using appropriate credentials and accept the subsequence Copyright Legal Notice.
2. Begin configuration by selecting Network Routing Policy from the left panel menu. A short procedure for configuring Network Routing Policy is shown on the right panel.
3. Add the SIP domain, for which the communications infrastructure will be authoritative, by selecting SIP Domains on the left panel menu and clicking the New button (not shown) to create a new SIP domain entry. The following screen will be shown after clicking New. Click Commit to save changes. Name Notes The authoritative domain name (e.g., avaya.com) Description for the domain (optional)
Note: Since the sample network does not deal with any foreign domains, no additional SIP Domains entry is needed.
4. A SIP Entity must be added to Avaya Aura Session Manager for each SIP-based telephony system supported by a SIP Trunk. To add a SIP Entity, select SIP Entities on the left panel menu and then click on the New button (not shown) on the right. Enter the following for each SIP Entity. Under General: Name FQDN or IP Address Type
Location Time Zone
An informative name (e.g., Avaya-S8500) IP address of the ASM or the signaling interface on the telephony system Session Manager for Avaya Aura Session Manager, CM for Avaya Aura Communication Manager, or Other for Cisco AS5400 The location that this SIP Entity belongs in Time zone for this location
The following screen shows the SIP Entity for Avaya AuraTM Communication Manager.
The following screen shows the SIP Entity for Cisco AS5400.
5. A SIP trunk between Avaya Aura Session Manager and a telephony system is described by an Entity link. To add an Entity Link, select Entity Links on the left panel menu and click on the New button on the right. Fill in the following fields in the new row that is displayed. Name SIP Entity 1 Port SIP Entity 2 Port Trusted Protocol An informative name Select SM1 Port number to which the other system sends its SIP requests The other SIP Entity for this link, created in Step 4 Port number to which the other system expects to receive SIP requests Whether to trust the other system Transport protocol to be used to send SIP requests
Click Commit to save changes. The following screen shows the Entity Links used in the sample network.
6. Before adding routing policies (see next step), time ranges must be defined during which the policies will be active. In the sample network, one policy was defined that would allow routing to occur at anytime. To add this time range, select Time Ranges from the left panel menu and then click New on the right. Fill in the following fields. Name Mo through Su Start Time End Time An informative name (e.g. Anytime) Check the box under each day of the week for inclusion Enter start time (e.g. 00:00 for start of day) Enter end time (e.g. 23:59 for end of day)
7. Create routing policies to direct how a call should be handled by the system. Two routing policies must be added; one for Communication Manager and one for AS5400. To add a routing policy, select Routing Policies on the left panel menu and then click on the New button (not shown) on the right. Under General Enter an informative name Under SIP Entity as Destination Click Select, and then select the appropriate SIP entity to which this routing policy applies Under Time of Day Click Add, and then select the time range configured in the previous step.
The following is screen shows the Routing Policy for Communication Manager.
The following is screen shows the Routing Policy for Cisco AS5400.
8. A dial pattern must be defined that will direct calls to the appropriate telephony system. In the sample network the 5-digit extension 21816 needs to be routed to Communication Manager and the 10-digit extension beginning with 732 needs to be routed to the Cisco AS5400. To add a dial pattern, select Dial Patterns on the left panel menu, and then click on the New button on the right (not shown). Under General Pattern Min Max Notes
Dialed number or prefix Minimum length of dialed number Maximum length of dialed number Comment on purpose of dial pattern
The following screen shows the dial pattern to Avaya Aura Communication Manager.
The following screen shows the dial pattern to Cisco AS5400.
7. Configure Cisco AS5400

The following sections provide annotated configuration steps for supporting voice/fax call routing via the Cisco AS5400 Universal Gateway. They are not a complete set of configuration steps for the product; only configuration commands relative to the sample configuration are shown. Access the AS5400 via telnet or the console interface.
7.1. Define Voice Services and Interfaces

Note that the voice configuration commands are global and apply to all interfaces and dial-peers
! voice service pots fax protocol t.38 g711ulaw ! voice service voip clid strip pi-restrict all fax protocol t.38 g711ulaw sip bind control source-interface FastEthernet0/1 bind media source-interface FastEthernet0/1 session transport tcp asymmetric payload dtmf
T.38 Fax (replace with fax protocol pass-through g711ulaw for voice mode Same as above for VoIP Restricted call support Bind signaling & meida to specific Ethernet port SIP signaling via TCP Allow different rcv/xmt DTMF payload
! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! interface FastEthernet0/1 ip address 10.1.2.40 255.255.255.0 no ip route-cache cef no ip route-cache duplex auto speed auto !
types Global Codec set to g.711mu-law and g729r8 for G.729A
7.2. Define PSTN Interface

The following commands configure the T1 PRI interface.
! isdn switch-type primary-ni voice call send-alert ! ! controller T1 2/0 framing esf linecode b8zs pri-group timeslots 1-24 ! ! interface Serial2/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn timer T306 30000 isdn timer T310 40000 isdn incoming-voice modem isdn T309-enable no cdp enable ! ! voice-port 2/0:D !
Set ISDN PRI switch protocol Send ALERT message (rather than PROGRESS) back on receipt of SETUP message. Configure T1 controller 2/0 Use all 23 bearer channels Configure PRI signaling channel (last channel) DISCONNECT timer CALL PROCEEDING timer Enable call clearing timer No Cisco Discovery Protocol Switch to voice-port config mode using PRI D
7.3. Define Call Routing

The following commands specify call routing to Session Manager. Note that the AS5400 will match a dialed number to the destination-pattern specified in the dial peers to determine which dial peer to use for routing the call. In the sample configuration, the AS5400 receives the last 5 digits of the called number (e.g., 21816) for incoming calls from the PSTN. For SIP trunks, the session target ipv4:10.1.2.170 command causes incoming calls to be routed to Session Manager.
! ! dial-peer voice 21816 voip service session destination-pattern 21816 progress_ind setup enable 1 voice-class codec 1 session protocol sipv2 session target ipv4:10.1.2.170 session transport tcp dtmf-relay rtp-nte ! ! ! dial-peer voice 732 pots service session destination-pattern 732....... direct-inward-dial port 2/0:D forward-digits 10 prefix *9 ! VoIP dial peer Inbound PSTN calls come into the AS5400 with this extension
SIP signaling Route to Avaya SM Use RFC 2833 for signaling DTMF Codec preferences (see Section 5, Sep 5 T1/PRI dial peer Outbound calls for area code 732 T1 signaling interface Forward AAA-NNN-XXXX Prefix *9 for outside calls
8. Verification
All features shown in Table 1 were tested using SIP trunks between Session Manager and AS5400 and Communication Manager shown in the sample configuration. The following steps can be used to verify/troubleshoot installations in the field. 1. Place a call from a PSTN phone to the DID that is mapped to an extension on Communication Manager. Use the list trace tac command to confirm that the call arrives on the configured SIP trunk in Communication Manager. Use a call that permits verification of DTMF support (e.g., let the call to go to voice mail). 2. Place a call from a Communication Manager extension to a PSTN phone. Use the list trace tac command to confirm that the call is routed outbound on the configured SIP trunk in Communication Manager. Use a call that permits verification of DTMF support (e.g., let the call to go to voice mail). 3. Use a network sniffer to verify that INVITE messages are properly routed between the Session Manager and Cisco AS5400. If DTMF is not successfully transmitted, check proper configuration of the Cisco AS5400 (see Section 7). Connect fax machines to a PSTN analog port and to an analog port on Avaya Media Gateways, as applicable. Confirm successful calls and accurate fax transmissions. If they fail, use the previous verification steps to verify that the calls are properly routed, and that T.38 fax mode is achieved, if applicable. Note that G.711 voice mode fax may not work in networks experiencing excessive packet loss and jitter.
8.1. Verify Avaya Aura Communication Manager

1. Verify the status of the SIP trunk group by using the status trunk n command, where n is the trunk group number being investigated. Verify that all trunks are in the in-service/idle state as shown below.
status trunk 25 TRUNK GROUP STATUS Member 0025/001 0025/002 0025/003 0025/004 0025/005 0025/006 0025/007 0025/008 0025/009 0025/010 Port T00007 T00008 T00009 T00010 T00011 T00012 T00013 T00014 T00015 T00016 Service State in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle Mtce Connected Ports Busy no no no no no no no no no no
2. Verify the status of the SIP signaling-group by using the status signaling-group n command, where n is the signaling group number being investigated. Verify that the signaling group is in the in-service state as shown below.
status signaling-group 25 STATUS SIGNALING GROUP Group ID: Group Type: Signaling Type: Group State: 25 sip facility associated signaling in-service Active NCA-TSC Count: 0 Active CA-TSC Count: 0
8.2. Verify Avaya Aura Session Manager

1. Expand the Session Manager field on the left panel menu and click on SIP Monitoring.
2. Select the SIP Entity Name entities, shown in the previous screen, and verify that the connection status is Up, as shown below for AS5400.
9. Conclusion
As illustrated in these Application Notes, Avaya AuraTM Session Manager can support SIP trunking to Cisco AS 5400 Universal Gateway for PRI connection to the PSTN. Basic calling, Hold, Conference, Transfer (Attended and Blind), Fax (T.38 and passthrough) as well as other supplementary feature such as Call Pickup, Call Park, are supported in this configuration. The following is a list of interoperability items to note: Call Forwarding from an Avaya telephone to another Avaya telephone is not support for incoming PSTN call. Cisco AS 5400 does not support call being forward from one Avaya telephone to another Avaya telephone. The Cisco AS5400 fail to respond to SIP signaling from Session Manager thus preventing the call from being established between the PSTN and Forward-to telephone.
10. Additional References

Product documentation for Avaya products may be found at http://support.avaya.com [1] [2] [3] [4] [5] Avaya AuraTM Session Manager Overview, Doc # 03-603323 Installing and Administering Avaya AuraTM Session Manager, Doc # 03-603324 Maintaining and Troubleshooting Avaya AuraTM Session Manager, Doc # 03603325 SIP Support in Avaya AuraTM Communication Manager Running on Avaya S8xxx Servers, Doc # 555-245-206, May 2009 Administering Avaya AuraTM Communication Manager, Doc # 03-300509, May 2009
Product documentation for Cisco Systems products may be found at http://www.cisco.com
2009 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 Solution & Interoperability Test Lab at interoplabnotes@list.avaya.com

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AL; Reviewed: SPOC 10/6/2009

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Figure 1: Sample Network Configuration

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3. Equipment and Software Validated

Table 1: SIP Telephony Feature Support

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5. Configure Avaya Aura Communication Manager

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change ip-codec-set 1 IP Codec Set Allow Direct-IP Multimedia? n

FAX Modem TDD/TTY Clear-channel

Mode t.38-standard off US n

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10. Configure a FEATURE ACCESS CODE to access ARS.

Percent Full: ANI Reqd n n n

DCS/ QSIG Intw n n n

IXC user user user

No. Numbering LAR Dgts Format Subaddress none none

13. Use the save translation command to save all changes.

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6. Configuring Avaya Aura Session Manager

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Location Time Zone

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The following screen shows the dial pattern to Cisco AS5400.

7. Configure Cisco AS5400

7.1. Define Voice Services and Interfaces

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types Global Codec set to g.711mu-law and g729r8 for G.729A

7.2. Define PSTN Interface

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7.3. Define Call Routing

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8.1. Verify Avaya Aura Communication Manager

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8.2. Verify Avaya Aura Session Manager

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10. Additional References

Product documentation for Cisco Systems products may be found at http://www.cisco.com

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