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Lync Direct SIP to Call Manager 8.x Can't dial in to conference RRS feed

  • Question

  • Hello,

    I've seen lots of posts on the CallManager to Lync integration, must be something stupid I'm missing, but the problem is as follows: I can dial-out from Lync to PSTN through CallManager, we're not using enterprise voice only trying to enable some Dial-In numbers for conference. The dial in successfully routes to Lync, in logs I can see invite to sip:+xxxx@IP_of_lync_std

    TL_INFO(TF_PROTOCOL) [1]0DC8.1868::03/31/2012-23:52:56.568.000079d5 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
    Trace-Correlation-Id: 3549442660
    Instance-Id: 00005992
    Direction: incoming
    Peer: IP_CallManager:42036
    Message-Type: request
    Start-Line: INVITE sip:+xxxxxxxx@IP_Lync_Std:5060 SIP/2.0
    From: <sip:+xxxxxxxxxxxxx@IP_CallManager>;tag=683~ebf3e994-f07f-4b37-8d5c-9a6ca2ad1e1f-30156396
    To: <sip:+xxxxxxxx@IP_Lync_Std>
    CSeq: 101 INVITE
    Call-ID: aad51300-f77198e7-1b-58017ac@IP_CallManager
    Via: SIP/2.0/TCP IP_CallManager:5060;branch=z9hG4bK1a31288700
    Date: Sat, 31 Mar 2012 23:53:11 GMT
    Supported: timer,resource-priority,replaces
    Min-SE: 1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Call-Info: <sip:IP_CallManager:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 2866090752-0000065536-0000000006-0092280748
    Session-Expires: 1800
    P-Asserted-Identity: <sip:+xxxxxxxxxxx@IP_CallManager>
    Remote-Party-ID: <sip:+xxxxxxxxxxx@IP_CallManager>;party=calling;screen=yes;privacy=off
    Contact: <sip:+xxxxxxxxxxx@IP_CallManager:5060;transport=tcp>
    Max-Forwards: 70
    Content-Type: application/sdp
    Content-Length: 317
    Message-Body: v=0
    o=CiscoSystemsCCM-SIP 683 1 IN IP4 IP_CallManager
    s=SIP Call
    c=IN IP4 IP_ISR
    b=TIAS:64000
    t=0 0
    m=audio 23020 RTP/AVP 0 8 15 18 4
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:15 G728/8000
    a=ptime:30
    a=rtpmap:4 G723/8000
    a=ptime:30
    a=rtpmap:18 G729/8000
    a=ptime:20

    then

    TL_INFO(TF_DIAG) [1]0DC8.1868::03/31/2012-23:52:56.568.00007dd9 (SIPStack,SIPAdminLog::TraceDiagRecord:SIPAdminLog.cpp(147))$$begin_record
    LogType: diagnostic
    Severity: information
    Text: Response successfully routed
    SIP-Start-Line: SIP/2.0 401 Unauthorized
    SIP-Call-ID: aad51300-f77198e7-1b-58017ac@IP_CallManager
    SIP-CSeq: 101 INVITE
    Peer: IP_CallManager:42036
    Data: destination="Unknown"
    $$end_record

    and call fails with SIP/4

    TL_INFO(TF_PROTOCOL) [1]0DC8.1868::03/31/2012-23:52:56.568.00007e0a (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
    Trace-Correlation-Id: 3549442660
    Instance-Id: 00005993
    Direction: outgoing;source="local"
    Peer: IP_CallManager:42036
    Message-Type: response
    Start-Line: SIP/2.0 401 Unauthorized
    From: <sip:+xxxx@IP_CallManager>;tag=683~ebf3e994-f07f-4b37-8d5c-9a6ca2ad1e1f-30156396
    To: <sip:+Dial-In@IP_Lync_Std>;tag=9FC9B8C043A1EDC06E9B923A8A7AEB4C
    CSeq: 101 INVITE
    Call-ID: aad51300-f77198e7-1b-58017ac@IP_CallManager
    Date: Sat, 31 Mar 2012 23:52:56 GMT
    WWW-Authenticate: NTLM realm="SIP Communications Service", targetname="FQDN_Lync_Std", version=4
    WWW-Authenticate: Kerberos realm="SIP Communications Service", targetname="sip/FQDN_Lync_Std", version=4
    WWW-Authenticate: TLS-DSK realm="SIP Communications Service", targetname="FQDN_Lync_Std", version=4, sts-uri="https://FQDN_Lync_Std:443/CertProv/CertProvisioningService.svc"
    Via: SIP/2.0/TCP IP_CallManager:5060;branch=z9hG4bK1a31288700;ms-received-port=42036;ms-received-cid=FD00
    Server: RTC/4.0
    Content-Length: 0
    Message-Body: –
    $$end_record

    My dial-in sip address is sip:+xxxx@sip_domain obviously, not sip:+xxxx@IP_Lync_Std and on Cisco (I'm not a CallManager expert though) I can only set destination as IP or server name "server.sipdomain" which is not my sip domain

    many thanks in advance!

    Monday, April 2, 2012 12:37 PM

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