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3CX SIP Trunk Cisco UCM Help

Discussion in '3CX Phone System - General' started by edujim, Oct 16, 2012.

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  1. edujim

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    Hello,

    I am fairly new to 3CX, and know some stuff in Cisco UCM, I know this topic has been brought up before in past Threads but i dont see any posts/replies to any of them after 2010-2011.

    I have successfully created a SIP Trunk between 3cx v11 to Cisco UCM 7.1.

    I've got two concerns:

    1. when making a call 3CX extension to Cisco extension, It only shows SIP Port # on the display. I dont see the 3CX extension that is calling on cisco phone. (Cisco to 3CX works).
    Here is the log:

    16-10-2012 12:17:29.366 Leg L:27.2[Line:10000>>1505] is terminated: Cause: BYE from PBX
    16-10-2012 12:17:29.366 [CM503008]: Call(C:27): Call is terminated
    16-10-2012 12:17:29.365 Leg L:27.1[Extn] is terminated: Cause: BYE from 10.5.80.110:51593
    16-10-2012 12:17:25.654 Currently active calls - 1: [27]
    16-10-2012 12:17:20.757 [CM503007]: Call(C:27): Line:10000>>1505 has joined, contact <sip:20000@10.178.50.2:5060>
    16-10-2012 12:17:20.755 [CM503007]: Call(C:27): Extn:6504 has joined, contact <sip:6504@10.5.80.110:51593>
    16-10-2012 12:17:20.753 L:27.2[Line:10000>>1505] has joined to L:27.1[Extn]
    16-10-2012 12:17:15.869 [CM505003]: Provider:[CCM] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-CUCM7.1] PBX contact: [sip:20000@172.17.5.233:5060;transport=TCP]
    16-10-2012 12:17:15.809 [CM503025]: Call(C:27): Calling T:Line:10000>>1505@[Dev:sip:20000@10.178.50.2:5060] for L:27.1[Extn]
    16-10-2012 12:17:15.757 [CM503027]: Call(C:27): From: Extn:6504 ("Eddie Jimenez" <sip:6504@172.17.5.233:5060>) to T:Line:10000>>1505@[Dev:sip:20000@10.178.50.2:5060]
    16-10-2012 12:17:15.757 [CM503004]: Call(C:27): Route 1: from L:27.1[Extn] to T:Line:10000>>1505@[Dev:sip:20000@10.178.50.2:5060]
    16-10-2012 12:17:15.757 Line limit check: Current # of calls for line Lc:10000(@CCM[<sip:20000@10.178.50.2:5060>]) is 0; limit is 24
    16-10-2012 12:17:15.757 Call(C:27): Call from Extn:6504 to 1505 matches outbound rule 'Extension to Cisco'
    16-10-2012 12:17:15.755 [CM503001]: Call(C:27): Incoming call from Extn:6504 to <sip:1505@172.17.5.233:5060>

    2. In order to make a successful call from Cisco to 3cx i had to add/change Source ID on 3cx
    Request Line URI: User Part Custom Field Custom Value "6504" which is my extension on 3cx
    based myself on this article: http://www.3cx.com/blog/docs/source-identification-issues/

    Here is Activity Log when it wasnt working

    15-10-2012 17:22:39.843 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    Invite-UNK Recv Req INVITE from 10.178.50.3:56350 tid=2d756f8e0c Call-ID=da0d3500-7c18cbb-2e-332b20a@10.178.50.3:
    INVITE sip:6504@172.17.5.233:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.178.50.3:5060;branch=z9hG4bK2d756f8e0c
    Max-Forwards: 70
    Contact: <sip:1505@10.178.50.3:5060;transport=tcp>
    To: <sip:6504@172.17.5.233>
    From: "Eduardo Jimenez"<sip:1505@10.178.50.3>;tag=29a56480-65c8-4771-b8ff-c11fbf058889-38300632
    Call-ID: da0d3500-7c18cbb-2e-332b20a@10.178.50.3
    CSeq: 101 INVITE
    Expires: 180
    Session-Expires: 1800
    Min-SE: 1800
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    Call-Info: <sip:10.178.50.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Date: Mon, 15 Oct 2012 22:22:51 GMT
    Supported: timer, resource-priority, replaces, Geolocation
    User-Agent: Cisco-CUCM7.1
    Allow-Events: presence, kpml
    P-Asserted-Identity: "Eduardo Jimenez" <sip:1505@10.178.50.3>
    Content-Length: 0
    Remote-Party-ID: "Eduardo Jimenez" <sip:1505@10.178.50.3>;party=calling;screen=yes;privacy=off

    Here is my question: what changes or additions do i need to make for the other 3cx extensions to work, when someone from cisco extension dials them.

    Any help is greatly appreciated
    Eddie
     
  2. donbru01

    donbru01 New Member

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    Have you create correct trusted sip trunk on Cisco .
    Have you create correct Traslation Pattern for incoming call on Cisco and correct Partition ?
     
  3. edujim

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    sip trunk has been created
    I didnt see any of the other threads mention that i needed a translation pattern
    i created a route pattern that sends all calls with extension 6XXX thru sip trunk
     
  4. RichardCrabb1

    RichardCrabb1 New Member

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    Hi,
    We had a project where we needed to changeover one phone at a time meaning that the old phone system had to be retained until all were changed over. To make calls to the old Asterisk based system we used a bridge. This might work with one end slave, the other master. Another method, if retaining trunks, is to use a constant trusted parameter such as the IP address of the other PBX.
     
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