Contents

Domain-Based Routing Support on the Cisco UBE

First Published: June 15, 2011

Last Updated: July 22, 2011

The Domain-based routing feature provides support for matching an outbound dial peer based on the domain name or IP address provided in the request URI of the incoming SIP message or an inbound dial peer.

Domain-based routing enables for calls to be routed on the outbound dialpeer based on the domain name or IP address provided in the request Uniform Resource Identifier (URI) of incoming Session IP message.

Feature Information for Domain-Based Routing Support on the Cisco UBE

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http:/​/​www.cisco.com/​go/​cfn . An account on Cisco.com is not required.

Table 1 Feature Information for Domain-Based Routing Support on the Cisco UBE

Feature Name

Releases

Feature Information

Domain Based Routing Support on the Cisco UBE

15.2(1)T

The domain-based routing enables for calls to be routed on the outbound dial peer based on the domain name or IP address provided in the request URI (Uniform Resource Identifier) of incoming SIP message.

The following commands were introduced or modified: call-route, voice-class sip call-route.

Domain Based Routing Support on the Cisco UBE

Cisco IOS XE Release 3.8S

The domain-based routing enables for calls to be routed on the outbound dial peer based on the domain name or IP address provided in the request URI (Uniform Resource Identifier) of incoming SIP message.

The following commands were introduced or modified: call-route, voice-class sip call-route.

Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental. © 2011 Cisco Systems, Inc. All rights reserved

Restrictions for Domain-Based Routing Support on the Cisco UBE

Domain-based routing support is available only for SIP-SIP call flows.

Information About Domain-Based Routing Support on the Cisco UBE

When a dial peer has an application configured as a session application, then only the user parameter of the request URI is used and is sent from the inbound SIP SPI to the application. The session application performs a match on an outbound dial peer based on the user parameter of the request URI sent from the inbound dial peer. In the figure below, 567 is the user portion of the request-URI that is passed from the inbound dial peer to the application and the matching outbound dial-peer found is 1000.



With the introduction of the domain-based routing feature, all parameters including the domain name of the request URI will be sent to the application and the outbound dial peer can be matched with any parameter. In Figure 1, when the domain name example.com is used to match an outbound dial peer the resulting dial peer is 2000. The call route url command is used for configuring domain-based routing.

How to Configure Domain-Based Routing Support on the Cisco UBE

Configuring Domain-Based Routing at Global Level

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    voice service voip

    4.    sip

    5.    call-route url

    6.    exit


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Device> enable
     

    Enables privileged EXEC mode.

    • Enter your password if prompted.

     
    Step 2 configure terminal


    Example:
    Device# configure terminal
     

    Enters global configuration mode.

     
    Step 3 voice service voip


    Example:
    Device(config)# voice service voip
     

    Enters voice service configuration mode.

     
    Step 4 sip


    Example:
    Device(conf-voi-serv)# sip
     

    Enters voice service SIP configuration mode.

     
    Step 5 call-route url


    Example:
    Device(conf-serv-sip)# call-route url


    Example:

     

    Routes calls based on the URL.

     
    Step 6 exit


    Example:
    Device(conf-serv-sip)# exit
     

    Exits the current mode.

     

    Configuring Domain-Based Routing at Dial Peer Level

    SUMMARY STEPS

      1.    enable

      2.    configure terminal

      3.    dial-peer voice dial-peer tag voip

      4.    voice-class sip call-route url

      5.    exit


    DETAILED STEPS
       Command or ActionPurpose
      Step 1 enable


      Example:
      Device> enable
       

      Enables privileged EXEC mode.

      • Enter your password if prompted.

       
      Step 2 configure terminal


      Example:
      Device# configure terminal
       

      Enters global configuration mode.

       
      Step 3 dial-peer voice dial-peer tag voip


      Example:
      Device(config)# dial-peer voice  2  voip
       

      Enter dial peer voice configuration mode.

       
      Step 4 voice-class sip call-route url


      Example:
      Device(config-dial-peer)#


      Example:

      Routes calls based on the URL

              		 
      Step 5 exit


      Example:
      Device(config-dial-peer)# exit
       

      Exits the current mode.

       

      Verifying and Troubleshooting Domain-Based Routing Support on the Cisco UBE

      SUMMARY STEPS

        1.    enable

        2.    debug ccsip all

        3.    debug voip dialpeer inout


      DETAILED STEPS
        Step 1   enable

        Enables privileged EXEC mode.



        Example:
        Device> enable
        Step 2   debug ccsip all

        Enables all SIP-related debugging.



        Example:
        Device# debug ccsip all
Received:
INVITE sip:5555555555@[2208:1:1:1:1:1:1:1118]:5060 SIP/2.0
Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1115]:5060;branch=z9hG4bK83AE3
Remote-Party-ID: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;party=calling;screen=no;privacy=off
From: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;tag=627460F0-1259
To: <sip:5555555555@[2208:1:1:1:1:1:1:1118]>
Date: Tue, 01 Mar 2011 08:49:48 GMT
Call-ID: B30FCDEB-431711E0-8EDECB51-E9F6B1F1@2208:1:1:1:1:1:1:1115
Supported: 100rel,timer,resource-priority,replaces
Require: sdp-anat
Min-SE:  1800
Cisco-Guid: 2948477781-1125585376-2396638033-3925258737
User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1298969388
Contact: <sip:2222222222@[2208:1:1:1:1:1:1:1115]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 495
v=0
o=CiscoSystemsSIP-GW-UserAgent 7880 7375 IN IP6 2208:1:1:1:1:1:1:1115
s=SIP Call
c=IN IP6 2208:1:1:1:1:1:1:1115
t=0 0
a=group:ANAT 1 2
m=audio 17836 RTP/AVP 0 101 19
c=IN IP6 2208:1:1:1:1:1:1:1115
a=mid:1                                                
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
m=audio 18938 RTP/AVP 0 101 19
c=IN IP4 9.45.36.111
a=mid:2                                                
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
“Received: 
INVITE sip:2222222222@[2208:1:1:1:1:1:1:1117]:5060 SIP/2.0
Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1116]:5060;branch=z9hG4bK38ACE
Remote-Party-ID: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;party=calling;screen=no;privacy=off
From: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;tag=4FE8C9C-1630
To: <sip:2222222222@[2208:1:1:1:1:1:1:1117]>;tag=1001045C-992
Date: Thu, 10 Feb 2011 12:15:08 GMT
Call-ID: 5DEDB77E-ADC11208-808BE770-8FCACF34@2208:1:1:1:1:1:1:1117
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1432849350-0876876256-2424621905-3925258737
User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1297340108
Contact: <sip:5555555555@[2208:1:1:1:1:1:1:1116]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 424
v=0
o=CiscoSystemsSIP-GW-UserAgent 8002 7261 IN IP6 2208:1:1:1:1:1:1:1116
s=SIP Call
c=IN IP6 2208:1:1:1:1:1:1:1116
t=0 0
m=image 17278 udptl t38
c=IN IP6 2208:1:1:1:1:1:1:1116
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy”
        Step 3   debug voip dialpeer inout

        The debug ccsip all and debug voip dialpeer inout commands can be entered in any order and any of the commands can be used for debugging depending on the requirement.



        Example:
        Displays information about the voice dial peers
Device# debug voip dialpeer inout

voip dialpeer inout debugging is on

        The following event shows the calling and called numbers:



        Example:
        *May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Calling Number=4085550111, Called Number=3600, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH

        The following event shows the incoming dial peer:



        Example:
        *May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100
*May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Calling Number=4085550111, Called Number=3600, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3600, Peer Info Type=DIALPEER_INFO_SPEECH
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3600
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)

        The following event shows the matched dial peers in the order of priority:



        Example:
           
List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=3600
     2: Dial-peer Tag=36

        Configuration Examples for Domain-Based Routing Support on the Cisco UBE

        Example Configuring Domain-Based Routing Support on the Cisco UBE

        The following example shows how to enable domain-based routing support on the Cisco UBE: 

        Device> enable
Device# configure terminal
Device(config)# voice service voip
Device(conf-voi-serv)# sip
Device(conf-serv-sip)# call-route url
Device(conf-serv-sip)# exit
Device(config)# dial-peer voice 2 voip
Device(config-dial-peer)# voice-class sip call-route url
Device(config-dial-peer)# exit
        Domain-Based Routing Support on the Cisco UBE

        Domain-Based Routing Support on the Cisco UBE

        First Published: June 15, 2011

        Last Updated: July 22, 2011

        The Domain-based routing feature provides support for matching an outbound dial peer based on the domain name or IP address provided in the request URI of the incoming SIP message or an inbound dial peer.

        Domain-based routing enables for calls to be routed on the outbound dialpeer based on the domain name or IP address provided in the request Uniform Resource Identifier (URI) of incoming Session IP message.

        Feature Information for Domain-Based Routing Support on the Cisco UBE

        The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

        Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http:/​/​www.cisco.com/​go/​cfn . An account on Cisco.com is not required.

        Table 1 Feature Information for Domain-Based Routing Support on the Cisco UBE

        Feature Name

        Releases

        Feature Information

        Domain Based Routing Support on the Cisco UBE

        15.2(1)T

        The domain-based routing enables for calls to be routed on the outbound dial peer based on the domain name or IP address provided in the request URI (Uniform Resource Identifier) of incoming SIP message.

        The following commands were introduced or modified: call-route, voice-class sip call-route.

        Domain Based Routing Support on the Cisco UBE

        Cisco IOS XE Release 3.8S

        The domain-based routing enables for calls to be routed on the outbound dial peer based on the domain name or IP address provided in the request URI (Uniform Resource Identifier) of incoming SIP message.

        The following commands were introduced or modified: call-route, voice-class sip call-route.

        Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental. © 2011 Cisco Systems, Inc. All rights reserved

        Restrictions for Domain-Based Routing Support on the Cisco UBE

        Domain-based routing support is available only for SIP-SIP call flows.

        Information About Domain-Based Routing Support on the Cisco UBE

        When a dial peer has an application configured as a session application, then only the user parameter of the request URI is used and is sent from the inbound SIP SPI to the application. The session application performs a match on an outbound dial peer based on the user parameter of the request URI sent from the inbound dial peer. In the figure below, 567 is the user portion of the request-URI that is passed from the inbound dial peer to the application and the matching outbound dial-peer found is 1000.



        With the introduction of the domain-based routing feature, all parameters including the domain name of the request URI will be sent to the application and the outbound dial peer can be matched with any parameter. In Figure 1, when the domain name example.com is used to match an outbound dial peer the resulting dial peer is 2000. The call route url command is used for configuring domain-based routing.

        How to Configure Domain-Based Routing Support on the Cisco UBE

        Configuring Domain-Based Routing at Global Level

        SUMMARY STEPS

          1.    enable

          2.    configure terminal

          3.    voice service voip

          4.    sip

          5.    call-route url

          6.    exit


        DETAILED STEPS
           Command or ActionPurpose
          Step 1 enable


          Example:
          Device> enable
           

          Enables privileged EXEC mode.

          • Enter your password if prompted.

           
          Step 2 configure terminal


          Example:
          Device# configure terminal
           

          Enters global configuration mode.

           
          Step 3 voice service voip


          Example:
          Device(config)# voice service voip
           

          Enters voice service configuration mode.

           
          Step 4 sip


          Example:
          Device(conf-voi-serv)# sip
           

          Enters voice service SIP configuration mode.

           
          Step 5 call-route url


          Example:
          Device(conf-serv-sip)# call-route url


          Example:

           

          Routes calls based on the URL.

           
          Step 6 exit


          Example:
          Device(conf-serv-sip)# exit
           

          Exits the current mode.

           

          Configuring Domain-Based Routing at Dial Peer Level

          SUMMARY STEPS

            1.    enable

            2.    configure terminal

            3.    dial-peer voice dial-peer tag voip

            4.    voice-class sip call-route url

            5.    exit


          DETAILED STEPS
             Command or ActionPurpose
            Step 1 enable


            Example:
            Device> enable
             

            Enables privileged EXEC mode.

            • Enter your password if prompted.

             
            Step 2 configure terminal


            Example:
            Device# configure terminal
             

            Enters global configuration mode.

             
            Step 3 dial-peer voice dial-peer tag voip


            Example:
            Device(config)# dial-peer voice  2  voip
             

            Enter dial peer voice configuration mode.

             
            Step 4 voice-class sip call-route url


            Example:
            Device(config-dial-peer)#


            Example:

            Routes calls based on the URL

                          		 
            Step 5 exit


            Example:
            Device(config-dial-peer)# exit
             

            Exits the current mode.

             

            Verifying and Troubleshooting Domain-Based Routing Support on the Cisco UBE

            SUMMARY STEPS

              1.    enable

              2.    debug ccsip all

              3.    debug voip dialpeer inout


            DETAILED STEPS
              Step 1   enable

              Enables privileged EXEC mode.



              Example:
              Device> enable
              Step 2   debug ccsip all

              Enables all SIP-related debugging.



              Example:
              Device# debug ccsip all
Received:
INVITE sip:5555555555@[2208:1:1:1:1:1:1:1118]:5060 SIP/2.0
Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1115]:5060;branch=z9hG4bK83AE3
Remote-Party-ID: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;party=calling;screen=no;privacy=off
From: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;tag=627460F0-1259
To: <sip:5555555555@[2208:1:1:1:1:1:1:1118]>
Date: Tue, 01 Mar 2011 08:49:48 GMT
Call-ID: B30FCDEB-431711E0-8EDECB51-E9F6B1F1@2208:1:1:1:1:1:1:1115
Supported: 100rel,timer,resource-priority,replaces
Require: sdp-anat
Min-SE:  1800
Cisco-Guid: 2948477781-1125585376-2396638033-3925258737
User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1298969388
Contact: <sip:2222222222@[2208:1:1:1:1:1:1:1115]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 495
v=0
o=CiscoSystemsSIP-GW-UserAgent 7880 7375 IN IP6 2208:1:1:1:1:1:1:1115
s=SIP Call
c=IN IP6 2208:1:1:1:1:1:1:1115
t=0 0
a=group:ANAT 1 2
m=audio 17836 RTP/AVP 0 101 19
c=IN IP6 2208:1:1:1:1:1:1:1115
a=mid:1                                                
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
m=audio 18938 RTP/AVP 0 101 19
c=IN IP4 9.45.36.111
a=mid:2                                                
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
“Received: 
INVITE sip:2222222222@[2208:1:1:1:1:1:1:1117]:5060 SIP/2.0
Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1116]:5060;branch=z9hG4bK38ACE
Remote-Party-ID: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;party=calling;screen=no;privacy=off
From: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;tag=4FE8C9C-1630
To: <sip:2222222222@[2208:1:1:1:1:1:1:1117]>;tag=1001045C-992
Date: Thu, 10 Feb 2011 12:15:08 GMT
Call-ID: 5DEDB77E-ADC11208-808BE770-8FCACF34@2208:1:1:1:1:1:1:1117
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1432849350-0876876256-2424621905-3925258737
User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1297340108
Contact: <sip:5555555555@[2208:1:1:1:1:1:1:1116]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 424
v=0
o=CiscoSystemsSIP-GW-UserAgent 8002 7261 IN IP6 2208:1:1:1:1:1:1:1116
s=SIP Call
c=IN IP6 2208:1:1:1:1:1:1:1116
t=0 0
m=image 17278 udptl t38
c=IN IP6 2208:1:1:1:1:1:1:1116
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy”
              Step 3   debug voip dialpeer inout

              The debug ccsip all and debug voip dialpeer inout commands can be entered in any order and any of the commands can be used for debugging depending on the requirement.



              Example:
              Displays information about the voice dial peers
Device# debug voip dialpeer inout

voip dialpeer inout debugging is on

              The following event shows the calling and called numbers:



              Example:
              *May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Calling Number=4085550111, Called Number=3600, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH

              The following event shows the incoming dial peer:



              Example:
              *May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100
*May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Calling Number=4085550111, Called Number=3600, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*May  1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3600, Peer Info Type=DIALPEER_INFO_SPEECH
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3600
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*May  1 19:32:11.735: //-1/6372E2598012/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)

              The following event shows the matched dial peers in the order of priority:



              Example:
                 
List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=3600
     2: Dial-peer Tag=36

              Configuration Examples for Domain-Based Routing Support on the Cisco UBE

              Example Configuring Domain-Based Routing Support on the Cisco UBE

              The following example shows how to enable domain-based routing support on the Cisco UBE: 

              Device> enable
Device# configure terminal
Device(config)# voice service voip
Device(conf-voi-serv)# sip
Device(conf-serv-sip)# call-route url
Device(conf-serv-sip)# exit
Device(config)# dial-peer voice 2 voip
Device(config-dial-peer)# voice-class sip call-route url
Device(config-dial-peer)# exit