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Problems with Lync 2013 incoming call. Terminate call after 3s RRS feed

  • Question

  • When I make an incoming call Lync server terminates the call in approx 3s. I get an invite on the client, if I pickup with client I have 2way audio. After 3s sec the call is ended by the Lync server. Does anyone have a idea what the solution could be? I attached the logs.

    INVITE sip:31208902151@<IP nr>:5060 SIP/2.0
    Via: SIP/2.0/TCP  <SIP trunk>;x-route-tag="cid:DMS-TRUNK@<SIP trunk>";branch=z9hG4bK1D7BA368E
    From: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    To: <sip:31208902151@<IP nr>>
    Date: Tue, 03 Sep 2013 11:04:42 GMT
    Call-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    Supported: timer,replaces
    Min-SE:  1800
    Cisco-Guid: 2011974096-331289059-3106734114-2438473394
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Remote-Party-ID: <sip:31206557575@<SIP trunk>>;party=calling;screen=yes;privacy=off
    Timestamp: 1378206282
    Contact: <sip:31206557575@<SIP trunk>:5060;transport=tcp>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 271


    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3073 4397 IN IP4 <SIP trunk>
    s=SIP Call
    c=IN IP4 <SIP trunk>
    t=0 0
    m=audio 16948 RTP/AVP 0 101
    c=IN IP4 <SIP trunk>
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=direction:passive
    SIP/2.0 100 Trying
    FROM: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    TO: <sip:31208902151@<IP nr>>
    CSEQ: 101 INVITE
    CALL-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    VIA: SIP/2.0/TCP <SIP trunk>;branch=z9hG4bK1D7BA368E;x-route-tag="cid:DMS-TRUNK@<SIP trunk>"
    CONTENT-LENGTH: 0
    TIMESTAMP: 1378206282


    SIP/2.0 183 Session Progress
    FROM: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    TO: <sip:31208902151@<IP nr>>;tag=e4df20beea;epid=252126EF62
    CSEQ: 101 INVITE
    CALL-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    VIA: SIP/2.0/TCP <SIP trunk>;branch=z9hG4bK1D7BA368E;x-route-tag="cid:DMS-TRUNK@<SIP trunk>"
    CONTACT: <sip:srv-w2k12-lync.qubenl.local:5060;transport=Tcp;maddr=<Lync nr>>
    CONTENT-LENGTH: 0
    ALLOW: CANCEL
    ALLOW: BYE
    ALLOW: UPDATE
    ALLOW: PRACK
    SERVER: RTCC/5.0.0.0 MediationServer


    SIP/2.0 180 Ringing
    FROM: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    TO: <sip:31208902151@<IP nr>>;tag=e4df20beea;epid=252126EF62
    CSEQ: 101 INVITE
    CALL-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    VIA: SIP/2.0/TCP <SIP trunk>;branch=z9hG4bK1D7BA368E;x-route-tag="cid:DMS-TRUNK@<SIP trunk>"
    CONTACT: <sip:srv-w2k12-lync.qubenl.local:5060;transport=Tcp;maddr=<Lync nr>>
    CONTENT-LENGTH: 0
    ALLOW: CANCEL
    ALLOW: BYE
    ALLOW: UPDATE
    ALLOW: PRACK
    SERVER: RTCC/5.0.0.0 MediationServer

    SIP/2.0 183 Session Progress
    FROM: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    TO: <sip:31208902151@<IP nr>>;tag=e4df20beea;epid=252126EF62
    CSEQ: 101 INVITE
    CALL-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    VIA: SIP/2.0/TCP <SIP trunk>;branch=z9hG4bK1D7BA368E;x-route-tag="cid:DMS-TRUNK@<SIP trunk>"
    CONTACT: <sip:srv-w2k12-lync.qubenl.local:5060;transport=Tcp;maddr=<Lync nr>>
    CONTENT-LENGTH: 0
    ALLOW: CANCEL
    ALLOW: BYE
    ALLOW: UPDATE
    ALLOW: PRACK
    SERVER: RTCC/5.0.0.0 MediationServer

    SIP/2.0 200 OK
    FROM: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    TO: <sip:31208902151@<IP nr>>;tag=e4df20beea;epid=252126EF62
    CSEQ: 101 INVITE
    CALL-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    VIA: SIP/2.0/TCP <SIP trunk>;branch=z9hG4bK1D7BA368E;x-route-tag="cid:DMS-TRUNK@<SIP trunk>"
    CONTACT: <sip:srv-w2k12-lync.qubenl.local:5060;transport=Tcp;maddr=<Lync nr>>
    CONTENT-LENGTH: 260
    SUPPORTED: timer
    SUPPORTED: 100rel
    CONTENT-TYPE: application/sdp
    ALLOW: ACK
    SERVER: RTCC/5.0.0.0 MediationServer
    Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
    Session-Expires: 1800;refresher=uas
    Min-SE: 1800

    v=0
    o=- 25 1 IN IP4 <Lync nr>
    s=session
    c=IN IP4 <Lync nr>
    b=CT:1000
    t=0 0
    m=audio 52178 RTP/AVP 0 101
    c=IN IP4 <Lync nr>
    a=rtcp:52179
    a=label:Audio
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    BYE sip:31206557575@<SIP trunk>:5060;transport=tcp SIP/2.0
    FROM: <sip:31208902151@<IP nr>>;epid=252126EF62;tag=e4df20beea
    TO: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    CSEQ: 1 BYE
    CALL-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP <Lync nr>:5060;branch=z9hG4bKec4b26d3
    CONTACT: <sip:srv-w2k12-lync.qubenl.local:5060;transport=Tcp;maddr=<Lync nr>>
    CONTENT-LENGTH: 0
    USER-AGENT: RTCC/5.0.0.0 MediationServer

    SIP/2.0 481 Call Leg/Transaction Does Not Exist
    Via: SIP/2.0/TCP <Lync nr>:5060;branch=z9hG4bKec4b26d3;received=<IP nr>
    From: <sip:31208902151@<IP nr>>;epid=252126EF62;tag=e4df20beea
    To: <sip:31206557575@<SIP trunk>>;tag=AE4A6220-1D52
    Call-ID: 77EE1E48-13BF11E3-B12BB02B-787E99BE@<SIP trunk>
    CSeq: 1 BYE
    Content-Length: 0

     

    Tuesday, September 3, 2013 1:16 PM

Answers

  • Hi,

    Does the outbound call work properly?

    Please make sure you have a Lync Site or Global dial pan or number manipulation on your gateway configured to handle the incoming call.

    The inbound calls must ultimately and exactly match a unique SIP number assigned to one of your users. Otherwise Lync will drop the call.


    Kent Huang
    TechNet Community Support

    • Marked as answer by Kent-Huang Tuesday, September 10, 2013 7:58 AM
    Thursday, September 5, 2013 6:59 AM