SIP Dial Rules Issue

Hi Guys,

    I am configuring a scenario where calls to E.164 numbers from Missed Calls should be routed. In order to avoid the interdigit timeout for SIP Phones, i configured a SIP Dial Rule just as the one Mark posted in his Blog article:

http://blog.ine.com/2011/02/04/ccieccvp-voice-trivia-contest-winner-gets-100-amazon-gift-card/

 

   After configuring and applying this SIP Dial Rules, i tested and Calls from Missed Calls in SIP Phones wer working just fine, but then i realized that calls from "Placed Calls" didn't.

 

Test #1 Procedure:

  • From a SIP Phone, configured with the SIP Dial Rule
  • Go off hook
  • Dial any allowed PSTN Phone number.
  • PSTN Phone Rings.
  • SIP Phone gets ringback tone.
  • PSTN goes off hook.
  • 2-way audio established.

Test #2 Procedure:

  • From the same SIP Phone, configured with the SIP Dial Rule
  • Go to "Placed Calls"
  • Highlight the previously (Successfully) called number.
  • PSTN Phone Rings.
  • SIP Phone goes Idle
  • PSTN keeps ringing.
  • PSTN goes off hook.
  • There is no audio.

 

Has anyone ran into this issue?

 

Regards,

Comments

  • Are there multiple paths the call can take to get to the PSTN?  It's behaving almost as if the two calls are taking different paths.  The lack of audio could be codec negotiation or some issue negotiating capabilities between the two end points.

    If it's straight SIP dial rule > Single route pattern > Single PSTN gateway then we'd know the call paths are the same, but I guess I'd suggest verifying paths before we look at this further.

    What protocol are you using between CUCM and the gateway?

  • Hi Matthew,

     

    • I tried with 2 different SIP Phones in 2 different sites. The First one, using a SIP-Trunk to it's GW, the other one is using H.323 to its GW.Both phones showed the same behavior. When i remove the SIP Dial Rule that get's fixed.
    • I do have multiple Paths. I will clean my config tonight to go deeper with this issue and post the reults.
    • I would not say that my issue is a one-way audio call, since the SIP Phone ends the call and goes Idle as soon as i press the Dial Softkey.
    • Regarding the Codec Negotiation. Is there a difference when using SIP Dial Rules?

     

    Thanks for taking the time.

     

    Kind Regards,

     

     

     

     

  • Matthew is correct -  there should be some problem with Codec neogtiation, and probably the call is getting different path each time.

    IP Phones Directories do not have any impact on RTP Stream at all.

    Please try to trace both calls, and see where it is going in both cases.

  • Hi Guys,

     

       Still having the same issue. I dialed 911 for test purposes, following are the debug ccsip messages from the Router:

    //////////////////////////////////////////SUCCESSFUL CALL///////////////////////////////////////////


    !!!!!!!!!!!!!!!!!!!! DIALED 911 FROM KEYPAD  !!!!!!!!!!!!!!!!!!!!!!



    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Date: Sat, 14 Apr 2012 00:55:44 GMT
    Call-Info: <sip:177.1.10.20:5060>;method="NOTIFY;Event=telephone-

    event;Duration=500"
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,

    NOTIFY
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    Allow-Events: presence, kpml
    P-Asserted-Identity: "CorpHQ Phone2" <sip:[email protected]>
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    Remote-Party-ID: "CorpHQ Phone2"

    <sip:[email protected]>;party=calling;screen=yes;privacy=off
    Content-Length: 0
    User-Agent: Cisco-CUCM7.0
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060;transport=tcp>
    Expires: 180
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK69193f9de5
    CSeq: 101 INVITE
    Session-Expires:  1800
    Max-Forwards: 69


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK69193f9de5
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    To: <sip:[email protected]>
    Date: Fri, 13 Apr 2012 17:56:24 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK69193f9de5
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    To: <sip:[email protected]>;tag=1F54BC0-6A1
    Date: Fri, 13 Apr 2012 17:56:24 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,

    NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 271

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 8803 7048 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 18300 RTP/AVP 0 8 18 19
    c=IN IP4 177.1.254.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:19 CN/8000


    !!!!!!!!!!!!!!!!!!!! 911 PHONE TAKES THE CALL  !!!!!!!!!!!!!!!!!!!!!!



    %ISDN-6-CONNECT: Interface Serial0/0/0:2 is now connected to 911 N/A
    %ISDN-6-CONNECT: Interface Serial0/0/0:2 is now connected to 911 N/A
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK69193f9de5
    From: "CorpHQ Phone2" <sip:[email protected]7.1.10.20>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    To: <sip:[email protected]>;tag=1F54BC0-6A1
    Date: Fri, 13 Apr 2012 17:56:24 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,

    NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 271

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 8803 7048 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 18300 RTP/AVP 0 8 18 19
    c=IN IP4 177.1.254.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:19 CN/8000

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Sat, 14 Apr 2012 00:55:44 GMT
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    Allow-Events: presence, kpml
    Content-Length: 154
    To: <sip:[email protected]>;tag=1F54BC0-6A1
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK6a4468081e
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 177.1.10.20
    s=SIP Call
    c=IN IP4 177.1.11.35
    t=0 0
    m=audio 27420 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=ptime:20



    !!!!!!!!!!!!!!!!!!!! EVERYTHING FINE, END CALL  !!!!!!!!!!!!!!!!!!!!!!



    %ISDN-6-DISCONNECT: Interface Serial0/0/0:2  disconnected from 911 , call lasted

    8 seconds
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK61300
    From: <sip:[email protected]>;tag=1F54BC0-6A1
    To: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    Date: Fri, 13 Apr 2012 17:56:34 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Timestamp: 1334339803
    CSeq: 101 BYE
    Reason: Q.850;cause=16
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:

    CorpHQ#SIP/2.0 200 OK
    Date: Sat, 14 Apr 2012 00:56:03 GMT
    From: <sip:[email protected]>;tag=1F54BC0-6A1
    Content-Length: 0
    To: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793622
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK61300
    CSeq: 101 BYE

    ////////////////////////////////////////////////////////////////////////////////////////////////////////////

     

     

    //////////////////////////////////////////UNSUCCESSFUL CALL///////////////////////////////////////////


    !!!!!!!!!!!!!!!!!!!! DIALED 911 FROM PLACED CALLS LIST  !!!!!!!!!!!!!!!!!!!!!!

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Date: Sat, 14 Apr 2012 01:00:05 GMT
    Call-Info: <sip:177.1.10.20:5060>;method="NOTIFY;Event=telephone-

    event;Duration=500"
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,

    NOTIFY
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    Allow-Events: presence, kpml
    P-Asserted-Identity: "CorpHQ Phone2" <sip:[email protected]>
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    Remote-Party-ID: "CorpHQ Phone2"

    <sip:[email protected]>;party=calling;screen=yes;privacy=off
    Content-Length: 0
    User-Agent: Cisco-CUCM7.0
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060;transport=tcp>
    Expires: 180
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    CSeq: 101 INVITE
    Session-Expires:  1800
    Max-Forwards: 69


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    To: <sip:[email protected]>
    Date: Fri, 13 Apr 2012 18:00:45 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    To: <sip:[email protected]>;tag=1F948D4-18F0
    Date: Fri, 13 Apr 2012 18:00:45 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY,

    INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 270

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 211 2820 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 16446 RTP/AVP 0 8 18 19
    c=IN IP4 177.1.254.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:19 CN/8000

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Date: Sat, 14 Apr 2012 01:00:05 GMT
    Call-Info: <sip:177.1.10.20:5060>;method="NOTIFY;Event=telephone-

    event;Duration=500"
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,

    NOTIFY
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    Allow-Events: presence, kpml
    P-Asserted-Identity: "CorpHQ Phone2" <sip:[email protected]>
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    Remote-Party-ID: "CorpHQ Phone2"

    <sip:[email protected]>;party=calling;screen=yes;privacy=off
    Content-Length: 0
    User-Agent: Cisco-CUCM7.0
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060;transport=tcp>
    Expires: 180
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7362e8e4dc
    CSeq: 101 INVITE
    Session-Expires:  1800
    Max-Forwards: 69


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7362e8e4dc
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    To: <sip:[email protected]>
    Date: Fri, 13 Apr 2012 18:00:45 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7362e8e4dc
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    To: <sip:[email protected]>;tag=1F94918-2149
    Date: Fri, 13 Apr 2012 18:00:45 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY,

    INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 271

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9300 4238 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 16386 RTP/AVP 0 8 18 19
    c=IN IP4 177.1.254.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:19 CN/8000

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    CANCEL sip:[email protected]:5060 SIP/2.0
    Date: Sat, 14 Apr 2012 01:00:05 GMT
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    Content-Length: 0
    To: <sip:[email protected]>
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    CSeq: 101 CANCEL
    Max-Forwards: 70


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    To: <sip:[email protected]>
    Date: Fri, 13 Apr 2012 18:00:46 GMT
    Call-ID: [email protected]
    CSeq: 101 CANCEL
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 487 Request Cancelled
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    To: <sip:[email protected]>;tag=1F948D4-18F0
    Date: Fri, 13 Apr 2012 18:00:46 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=16
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Date: Sat, 14 Apr 2012 01:00:05 GMT
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793624
    Allow-Events: presence, kpml
    Content-Length: 0
    To: <sip:[email protected]>;tag=1F948D4-18F0
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7267d39b8f
    CSeq: 101 ACK
    Max-Forwards: 70


    !!!!!!!!!!!!!!!!!!!! CALLING PHONE GOES IDLE !!!!!!!!!!!!!!!!!!!!

    !!!!!!!!!!!!!!!!!!!! 911 PHONE KEEPS RINGING !!!!!!!!!!!!!!!!!!!!

    !!!!!!!!!!!!!!!!!!!! 911 PHONE TAKES THE CALL!!!!!!!!!!!!!!!!!!!!



    %ISDN-6-CONNECT: Interface Serial0/0/0:1 is now connected to 911 N/A
    %ISDN-6-CONNECT: Interface Serial0/0/0:1 is now connected to 911 N/A
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK7362e8e4dc
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    To: <sip:[email protected]>;tag=1F94918-2149
    Date: Fri, 13 Apr 2012 18:00:45 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY,

    INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 271

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9300 4238 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 16386 RTP/AVP 0 8 18 19
    c=IN IP4 177.1.254.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:19 CN/8000

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Sat, 14 Apr 2012 01:00:05 GMT
    From: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    Allow-Events: presence, kpml
    Content-Length: 154
    To: <sip:[email protected]>;tag=1F94918-2149
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.10.20:5060;branch=z9hG4bK742bcd99f
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 177.1.10.20
    s=SIP Call
    c=IN IP4 177.1.11.35
    t=0 0
    m=audio 21284 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=ptime:20





    !!!!!!!!!!!!!!!!!!!! THERE IS NO AUDIO    !!!!!!!!!!!!!!!!!!!!


    !!!!!!!!!!!!!!!!!!!!!!!!!!  END CALL      !!!!!!!!!!!!!!!!!!!!!!






    %ISDN-6-DISCONNECT: Interface Serial0/0/0:1  disconnected from 911 , call lasted 11

    seconds
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK7611
    From: <sip:[email protected]>;tag=1F94918-2149
    To: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    Date: Fri, 13 Apr 2012 18:01:10 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Timestamp: 1334340081
    CSeq: 101 BYE
    Reason: Q.850;cause=16
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:

    CorpHQ#SIP/2.0 200 OK
    Date: Sat, 14 Apr 2012 01:00:41 GMT
    From: <sip:[email protected]>;tag=1F94918-2149
    Content-Length: 0
    To: "CorpHQ Phone2" <sip:[email protected]>;tag=06bb5c4b-f416-48f6-9225-

    822962b6cef1-35793626
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK7611
    CSeq: 101 BYE

    ////////////////////////////////////////////////////////////////////////////////////////////////////////////

    Regards,

     

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