Problem with Outgoing Calls with local SIP provider

Discussion in '3CX Phone System - General' started by erling, Dec 21, 2011.

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  1. erling

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    I'm having problems setting up trunks with a local SIP Provider here in Austin, TX (DoubleHorn Communications)

    The trunks register, but will only accept incoming calls (no outgoing)

    Here's the log from 3CX when a call is attempted

    15:05:04.734 [CM503016]: Call(65): Attempt to reach <sip:1512XXXXXXX@10.20.126.40:5060> failed. Reason: Not Found
    15:05:04.734 [CM503003]: Call(65): Call to sip:1512XXXXXXX@69.94.224.2:5060 has failed; Cause: 404 Not Found; from IP:69.94.224.2:5060
    15:05:04.624 [CM503025]: Call(65): Calling VoIPline:1512XXXXXXX@(Ln.10008@DH Test)@[Dev:sip:5125014229@69.94.224.2:5060]
    15:05:04.562 [CM503004]: Call(65): Route 1: VoIPline:1512XXXXXXX@(Ln.10008@DH Test)@[Dev:sip:5125014229@69.94.224.2:5060]
    15:05:04.562 [CM503010]: Making route(s) to <sip:1512XXXXXXX@10.20.126.40:5060>
    15:05:04.562 [CM505001]: Ext.10: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 6.0.19920.0] PBX contact: [sip:10@127.0.0.1:5060]
    15:05:04.562 [CM503001]: Call(65): Incoming call from Ext.10 to <sip:1512XXXXXXX@10.20.126.40:5060>

    They have sent me a log from their end on how they see the invite

    INVITE sip:1512XXXXXXX@69.94.224.2:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.20.126.40:5060;branch=z9hG4bK-d8754z-d821b65ebf6f106f-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:5125014229@24.153.178.146:5060>
    To: <sip:1512XXXXXXX@69.94.224.2:5060>
    From: "Murray Messelt"<sip:5125014229@69.94.224.2:5060>;tag=71758d02
    Call-ID: MWQ5YTIwM2IzYTkxMWZiNGVmYjk5YzJhM2Q3YjhhM2Y.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.22042.0
    Content-Length: 283
    Remote-Party-ID: "Murray Messelt"<sip:Murray%20Messelt@69.94.224.2:5060>;party=calling
    v=0
    o=3cxPS 317173268480 122859552769 IN IP4 24.153.178.146
    s=3cxPS Audio call
    c=IN IP4 24.153.178.146
    t=0 0
    m=audio 9004 RTP/AVP 0 8 3 101
    c=IN IP4 24.153.178.146
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    SIP/2.0 100 Trying
    To: <sip:1512XXXXXXX@69.94.224.2:5060>
    From: "Murray Messelt"<sip:5125014229@69.94.224.2:5060>;tag=71758d02
    Via: SIP/2.0/UDP 10.20.126.40:5060;branch=z9hG4bK-d8754z-d821b65ebf6f106f-1---d8754z-
    Call-ID: MWQ5YTIwM2IzYTkxMWZiNGVmYjk5YzJhM2Q3YjhhM2Y.
    CSeq: 1 INVITE
    Content-Length: 0
    SIP/2.0 404 Not Found
    To: <sip:1512XXXXXXX@69.94.224.2:5060>;tag=451f0e62
    From: "Murray Messelt"<sip:5125014229@69.94.224.2:5060>;tag=71758d02
    Via: SIP/2.0/UDP 10.20.126.40:5060;branch=z9hG4bK-d8754z-d821b65ebf6f106f-1---d8754z-
    Call-ID: MWQ5YTIwM2IzYTkxMWZiNGVmYjk5YzJhM2Q3YjhhM2Y.
    CSeq: 1 INVITE
    Server: DHPS-P900 v1.3.4-005
    Content-Length: 0

    Any ideas?

    Thanks in advance,

    Murray
     
  2. eagle2

    eagle2 Well-Known Member

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    Have you configured outgoing rule ?

    Regards
     
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  3. erling

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    Yes - just to pick that trunk when I call from a specific extension
     
  4. leejor

    leejor Well-Known Member

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    What is their explanation as to why it isn't going through???

    This looks like they don't know what to do with the digits you are sending them.
     
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