3cx MP-118 Speech Server

Discussion in '3CX Phone System - General' started by pranaywilson, Feb 18, 2008.

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  1. pranaywilson

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    I have a SIP Phone registered as extension 100 on the 3cx system.
    I also have all ports of a MP-118 gateway registered as lines of the 3cx system (10000 - 10007)
    All the lines are successfully registered.

    I have a speech application which receives the call from the gateway. After some speech activities, the speech application blind transfers to the SIP phone.
    The uri used for the blind transfer is 100@3cx.
    On receiving the REFER message the gateway sends an INVITE to the 3cx system, upon which the 3cx requests for authentication. Now the port on which the call has been received has user name: 10000 password: 10000, but the line I want to call is 100. So the gateway provides with the credentials configured for its port (10000), but the extension is 100. Therefore the authentication fails.
    The logs are shown below.

    1:29:43.046 AuthMgr::eek:nAuthFailure [CM102001]: Authentication failed for SipReq: INVITE 100@192.168.1.1 tid=ac892143327 cseq=INVITE contact=9548389518@192.168.1.9 / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
    11:29:42.750 evt::CheckAuth::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:100@192.168.1.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.9;branch=z9hG4bKac892143327
    Max-Forwards: 70
    Contact: [sip:9548389518@192.168.1.9]
    To: [sip:100@192.168.1.1]
    From: "ORDER CATCHER"[sip:9548389518@192.168.1.1:5062];tag=1c891059758
    Call-ID: 89105936031200020546@192.168.1.9
    CSeq: 2 INVITE
    Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
    Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="12847825782:370101bb130a6161707d4b9c90d2821f",uri="sip:100@192.168.1.1",algorithm=MD5,response="1257de4950387160c6238f0dc6ae6161"
    Supported: em, timer, replaces, path, resource-priority
    User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.017.001
    Referred-By: [sip:10000@192.168.1.1]
    Content-Length: 0
    Remote-Party-ID: "ORDER CATCHER" [sip:9548389518@192.168.1.1];party=calling;privacy=off;screen=no;screen-ind=0;npi=0;ton=0, [sip:100@192.168.1.1];party=called;npi=0;ton=0

    11:29:42.750 evt::CheckAuth::not_handled [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE 100@192.168.1.1 tid=ac892143327 cseq=INVITE contact=9548389518@192.168.1.9 / 2 from(wire)
    11:29:42.328 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:100@192.168.1.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.9;branch=z9hG4bKac891067054
    Max-Forwards: 70
    Contact: [sip:9548389518@192.168.1.9]
    To: [sip:100@192.168.1.1]
    From: "ORDER CATCHER"[sip:9548389518@192.168.1.1:5062];tag=1c891059758
    Call-ID: 89105936031200020546@192.168.1.9
    CSeq: 1 INVITE
    Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
    Supported: em, timer, replaces, path, resource-priority
    User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.017.001
    Referred-By: [sip:10000@192.168.1.1]
    Content-Length: 0
    Remote-Party-ID: "ORDER CATCHER" [sip:9548389518@192.168.1.1];party=calling;privacy=off;screen=no;screen-ind=0;npi=0;ton=0, [sip:100@192.168.1.1];party=called;npi=0;ton=0

    11:29:42.328 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE 100@192.168.1.1 tid=ac891067054 cseq=INVITE contact=9548389518@192.168.1.9 / 1 from(wire)
     
  2. pranaywilson

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    bump?

    Any suggestions?
     
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