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Siemens <-> 3CX SIP Trunk Inbound CID problems

Discussion in '3CX Phone System - General' started by gogusrl, Sep 11, 2017.

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  1. gogusrl

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    I'm trying to connect a Siemens HiPath 3770 to a 3CX server via a SIP Trunk (no auth). Outgoing works great, incoming not so much.

    IP of 3CX : 192.168.1.168
    IP of Siemens : 192.168.1.101

    Call Source identification is enabled with Contact "Host Part' -> "GWHostPort", Use both is disabled (if I enable it I get "Unidentified incoming call. Review INVITE and adjust source identification").

    There's 2 scenarios :

    1. Inbound parameters / Caller Number/Name Field Mapping all at Leave Default Value. With these settings the call will ignore all my Inbound Rules and is always routed to the Catch All destination I have setup in Trunk Setup.

    Code:
    09/11/2017 4:24:05 PM - [CM503010]: Call(C:192): Making route(s) from Line:10002<<728XXXXX to <sip:100@192.168.1.168:5060>
    09/11/2017 4:24:05 PM - [CM505003]: Provider:[Siemens] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [HiPath 3000 V8 M5T SIP Stack/4.0.26.26] PBX contact: [sip:10002@192.168.1.168:5060]
    09/11/2017 4:24:05 PM - [CM500002]: Call(C:192): Info on incoming INVITE from Line:10002<<728XXXXXX:
    Invite-IN Recv Req INVITE from 192.168.1.101:5060 tid=a97be6d5bb537f176.b965d8ede5653a1a9 Call-ID=1d850691a2a21a43:
    INVITE sip:110@192.168.1.168:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bKa97be6d5bb537f176.b965d8ede5653a1a9;rport=5060
    Max-Forwards: 70
    Contact: <sip:728XXXX@192.168.1.101:5060>
    To: <sip:110@192.168.1.168>
    From: 728XXXXXX<sip:728XXXXXX@192.168.1.101>;tag=428a04e3a0
    Call-ID: 1d850691a2a21a43
    CSeq: 22144 INVITE
    Session-Expires: 1800
    Min-SE: 90
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE
    Content-Type: application/sdp
    Supported: timer
    User-Agent: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
    P-Asserted-Identity: <sip:728XXXXX@192.168.1.101>
    Content-Length: 362
    
    v=0
    o=MxSIP 0 627345659 IN IP4 192.168.1.101
    s=SIP Call
    c=IN IP4 192.168.1.101
    t=0 0
    m=audio 29100 RTP/AVP 8 0 18 4 98 99
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:98 telephone-event/8000
    a=rtpmap:99 red/8000
    a=silenceSupp:off - - - -
    a=fmtp:4 annexa=no
    a=fmtp:98 0-15
    a=fmtp:99 98
    a=sendrecv
    09/11/2017 4:24:05 PM - [CM503001]: Call(C:192): Incoming call from Line:10002<<728XXXXX to <sip:100@192.168.1.168:5060>
    09/11/2017 4:24:05 PM - Line limit check: Current # of calls for line Lc:10002(@Siemens[<sip:10002@192.168.1.101:5060>]) is 1; limit is 10



    2. Inbound parameters -> Caller Number / Name Field Mapping -> "CallerNum" caller's number -> Set to Request Line URI: User Part. The call follows the Inbound Rules and delivers the call to the right extension but the caller ID shows up as CID : DID so if I'm calling from 11111111 to DID 222 in the 3CX client I see a call from "1111111 222" and I can't call it back.

    Code:
    09/11/2017 4:33:09 PM - [Flow] Call(C:193): has built target endpoint: Extn:311 for call from L:193.1[Line:10002<<110]
    09/11/2017 4:33:09 PM - [Flow] Target endpoint for 311 is Extn:311
    09/11/2017 4:33:09 PM - [CM503010]: Call(C:193): Making route(s) from Line:10002<<110 to <sip:311@192.168.1.168:5060>
    09/11/2017 4:33:09 PM - [CM505003]: Provider:[Siemens] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [HiPath 3000 V8 M5T SIP Stack/4.0.26.26] PBX contact: [sip:10002@192.168.1.168:5060]
    09/11/2017 4:33:09 PM - [CM500002]: Call(C:193): Info on incoming INVITE from Line:10002<<110:
    Invite-IN Recv Req INVITE from 192.168.1.101:5060 tid=f47658eb712e4bb3e.4caae0eea1fc73d94 Call-ID=1d82ec77fc16adbe:
    INVITE sip:110@192.168.1.168:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bKf47658eb712e4bb3e.4caae0eea1fc73d94;rport=5060
    Max-Forwards: 70
    Contact: <sip:728XXXXXX@192.168.1.101:5060>
    To: <sip:110@192.168.1.168>
    From: 728XXXXXX<sip:728XXXXXX@192.168.1.101>;tag=e01523c3e9
    Call-ID: 1d82ec77fc16adbe
    CSeq: 2277 INVITE
    Session-Expires: 1800
    Min-SE: 90
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE
    Content-Type: application/sdp
    Supported: timer
    User-Agent: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
    P-Asserted-Identity: <sip:728XXXXXX@192.168.1.101>
    Content-Length: 362
    
    v=0
    o=MxSIP 0 999390257 IN IP4 192.168.1.101
    s=SIP Call
    c=IN IP4 192.168.1.101
    t=0 0
    m=audio 29100 RTP/AVP 8 0 18 4 98 99
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:98 telephone-event/8000
    a=rtpmap:99 red/8000
    a=silenceSupp:off - - - -
    a=fmtp:4 annexa=no
    a=fmtp:98 0-15
    a=fmtp:99 98
    a=sendrecv
    09/11/2017 4:33:09 PM - [CM503001]: Call(C:193): Incoming call from Line:10002<<110 to <sip:311@192.168.1.168:5060>
    09/11/2017 4:33:09 PM - Line limit check: Current # of calls for line Lc:10002(@Siemens[<sip:10002@192.168.1.101:5060>]) is 1; limit is 10
    Please let me know if there's a need for additional information and thank you for taking a look at this (whoever you are).
     
  2. leejor

    leejor Well-Known Member

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    Are you using a Bridge Trunk, or a generic SIP trunk?
     
  3. gogusrl

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    I'm using a Generic SIP Trunk. Not sure what a Bridge Trunk is.
     
  4. leejor

    leejor Well-Known Member

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