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OCS drops incoming call after 30+ seconds RRS feed

  • Question

  •  Hi

    I've got a setup with OCS server , Mediation server , Opensips and Trixbox.

    When a call comes in from the outside I'm able to answer it on both the OC software and the snom (OCS) phone.  Unfortunately the call only lasts about 30+ seconds then drops.

    If I call out using the same system it has no problems.   I've tried looking at the logs but don't see anything unusual and no errors.  Then I get this message:

    TL_INFO(TF_PROTOCOL) [0]12C8.1258::12/19/2008-12:36:59.377.0000048c (SIPStack,SIPAdminLog::TraceProtocolRecord:1224.idx(122))$$begin_record

    Instance-Id: 000047E3

    Direction: incoming

    Peer: OCSMediator.xxx:1092

    Message-Type: request

    Start-Line: BYE sip:david@192.168.0.49:2067;transport=tls;line=rf6rf6pj;ms-received-cid=1A00 SIP/2.0

    From: <sip:021xxx;phone-context=Local@xxx;user=phone>;epid=D30AE2E34E;tag=1929b6c751

    To: <sip:+820@xxx;user=phone>;tag=r267eq194f

    CSeq: 122 BYE

    Call-ID: c8e4bce9-1801-4640-a7fc-f2aa86f4b2b5

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TLS 192.168.0.7:1092;branch=z9hG4bK987b3ce8

    ROUTE: <sip:OCS2007.xxx:5061;transport=tls;ms-role-rs-to;ms-opaque=aaDke_lWz25Aq9bkpFXRhU-QAA;lr>

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

    Message-Body:

    $$end_record

    Friday, December 19, 2008 1:08 PM

Answers

  • Problem solved seems like I needed nat enabled on the asterisk trunk to opensips.
    • Marked as answer by dkpeall Monday, December 22, 2008 11:52 AM
    Monday, December 22, 2008 11:52 AM

All replies

  • Check if there is a firewall somewhere in between
    - Belgian Exchange Community : http://www.pro-exchange.be -
    Friday, December 19, 2008 8:49 PM
  • You should use Logging Tool on Mediation Server and Snooper.exe from Res.Kit to analyze it. See if you get 200 OK from Mediation server. You may try it without the option "user=phone" on incoming INVITE. I have seen similar behaviour on that. And you should normalize both calling and called number to E.164 format.

    Johann Deutinger | MCTS Exchange 2007 / OCS 2007
    Saturday, December 20, 2008 9:59 AM
  • Yes the invite is there:

    One in each direction and then an ACK in each direction then nothing till 32 seconds late the BYE in each direction.

     TL_INFO(TF_PROTOCOL) [0]12C8.1258::12/19/2008-12:36:27.110.000002e6 (SIPStack,SIPAdminLog::TraceProtocolRecord:1224.idx(122))$$begin_record
    Instance-Id: 000047C1
    Direction: incoming
    Peer: 192.168.0.49:2067
    Message-Type: response
    Start-Line: SIP/2.0 200 Ok
    From: <sip:021xxx;phone-context=Local@xxx;user=phone>;tag=1929b6c751;epid=D30AE2E34E
    To: <sip:+820@xxx;user=phone>;tag=r267eq194f;epid=00041327B18A
    CSeq: 121 INVITE
    Call-ID: c8e4bce9-1801-4640-a7fc-f2aa86f4b2b5
    Via: SIP/2.0/TLS 192.168.0.48:5061;branch=z9hG4bK9E9AABC3.8B0E58E7;branched=FALSE;ms-internal-info="dbZDojeZQuDBI9ei3FU3A5unoCdRWLDljnaHa5CAAA"
    Via: SIP/2.0/TLS 192.168.0.7:1092;branch=z9hG4bKd3e4122c;ms-received-port=1092;ms-received-cid=3E00
    Record-Route: <sip:OCS2007.xxx:5061;transport=tls;ms-role-rs-from;lr;ms-identity=Bij64nqJFIx_iRSMaah9IP887lFAtP1dz_ingxWs-pNhyLDljn3GoQxAAA;ms-route-sig=aa8Sk4BJrt8fQDnNr6gMrz45hA9cmLDljn3GoQxAAA>;tag=27DCC02ECDC13AFE9B548F5757AA96EC
    Contact: <sip:david@192.168.0.49:2067;transport=tls;line=rf6rf6pj>;reg-id=1;proxy=replace
    User-Agent: snom320/r73ocs-wbc1119
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Authorization: NTLM qop="auth", realm="SIP Communications Service", opaque="42FCA9D1", targetname="OCS2007.xxx", crand="3d6d9870", cnum="1539", response="0100000021691effb28b186994d36ee0"
    Content-Type: application/sdp
    Content-Length: 606
    Message-Body: v=0
    o=root 1881651990 1881651992 IN IP4 192.168.0.49
    s=call
    c=IN IP4 192.168.0.49
    t=0 0
    m=audio 56238 RTP/SAVP 0 101
    a=rtpmap:0 pcmu/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:QGvvCA1Ye9ktSHzN7uaHuoKYIqmvJQ9lwCl78jNf|2^31|1:1
    a=rtcp:56239
    a=candidate:MDd1M3RoMmlvMGVqcWRsamFmYmJ4bmxpam90Y2V2Z2U 1 bnUxZ3ZuaTRveG5ldThoNA UDP 1 192.168.0.49 56238
    a=candidate:MDd1M3RoMmlvMGVqcWRsamFmYmJ4bmxpam90Y2V2Z2U 2 bnUxZ3ZuaTRveG5ldThoNA UDP 1 192.168.0.49 56239
    a=remote-candidate:VNmTBo2nKZHc391WUr5Rgcoj1UeitM62IVGoC9RjJDY
    a=sendrecv
    $$end_record
    Monday, December 22, 2008 6:33 AM
  • ok here is log from mediation server and the isAnswer is false ? On an outgoing call this is true and the call does not drop ? how do i remove the user=phone ?

    Thanks for the help

    TL_INFO(TF_COMPONENT) [0]0FF0.0E44::12/22/2008-07:12:32.557.00000032 (MediationServer,ProxyStream.ChangeStreamFlow:816.idx(1420))( 0311C5EC )$$START-MEDIATIONSERVER

    MediationCall: f1d4814813fa45548f52caf755208803

    CallId: a2882695-6b73-4d42-8598-0ad26bfc6514

    From: sip:0216838719;phone-context=Local@ct.esn.org.za;user=phone

    To: sip:+820@ct.esn.org.za;user=phone

    Direction: Inbound

    Start-Line: new ProxyStream direction is: Sendrecv , isAnswer: False

    $$END-MEDIATIONSERVER

    Monday, December 22, 2008 7:17 AM
  • Problem solved seems like I needed nat enabled on the asterisk trunk to opensips.
    • Marked as answer by dkpeall Monday, December 22, 2008 11:52 AM
    Monday, December 22, 2008 11:52 AM