404 Not Found for outgoing ISDN calls via Patton SN4634

Discussion in '3CX Phone System - General' started by Banana Jack, Apr 2, 2008.

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  1. Banana Jack

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    Hi all

    Everything is working with my new 3CX except I can't make outgoing ISDN calls through my Patton SN4634. Incoming and outgoing calls work through VOIP (sipgate) and incoming calls work through ISDN, but not outgoing through ISDN. Here is my server log:

    22:40:29.682 Call::Terminate [CM503008]: Call(32): Call is terminated
    22:40:29.650 Call::RouteFailed [CM503014]: Call(32): Attempt to reach [sip:01234567890@192.168.1.10;user=phone] failed. Reason: Not Found
    22:40:29.650 CallLeg::eek:nFailure [CM503003]: Call(32): Call to sip:01234567890@192.168.1.4:5062 has failed; Cause: 404 Not Found; from IP:192.168.1.4:5062
    22:40:29.588 CallLeg::eek:nFailure [CM503003]: Call(32): Call to sip:01234567890@192.168.1.4:5060 has failed; Cause: 404 Not Found; from IP:192.168.1.4:5060
    22:40:29.510 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(32): Calling: PSTNline:10001@[Dev:sip:10001@192.168.1.4:5060, Dev:sip:10002@192.168.1.4:5062]
    22:40:29.479 CallCtrl::eek:nIncomingCall [CM503001]: Call(32): Incoming call from Ext.10 to [sip:01234567890@192.168.1.10;user=phone]

    192.168.1.4 is my Patton SN4634
    192.168.1.10 is my 3CX server

    I loaded the config file from 3CX into the Patton but I think I must need to make some more configuration changes - can anyone please help me? I am only using 1 of the unit's 3 ISDN2e ports, if it makes a difference (BRI 0/0).

    Thanks for any help! I think this will be the last hurdle to deploying the whole system.

    Glenn
     
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  2. Banana Jack

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    I'm still struggling with this one. If anyone can tell me why 3CX can't make outgoing calls through my Patton, I'd be very grateful to know! Here is a log from the Patton itself when I try to make a call.

    Many thanks
    Glenn
    --
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2008.04.07 17:58:42 =~=~=~=~=~=~=~=~=~=~=~=

    login: admiinistrator
    password:
    192.168.1.4>enable
    192.168.1.4#debug call-router
    192.168.1.4#debug call-control
    192.168.1.4#debug gateway sip signaling detail 3
    192.168.1.4#debug gateway sip transport detail 3
    192.168.1.4#debug isdn error
    192.168.1.4#debug isdn event 0 0 layer3
    192.168.1.4#16:59:22 SIP_TR> [GW] < Stack: INVITE sip:01653669000@192.168.1.4:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d87543-20496a4d063a4e41-1--d87543-;rport
    Max-Forwards: 70
    Contact: <sip:10@192.168.1.10:5060>
    To: <sip:01653669000@192.168.1.4:5060>
    From: "10"<sip:10@192.168.1.10:5060>;tag=33278d71
    Call-ID: ZGI1NDdmNWZiMGQzM2IyMzIxOWY2MGYzZWI5ZWQ2NGY.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
    Content-Type: application/sdp
    User-Agent: 3CXPhoneSystem 5.1.4287.0
    Content-Length: 493

    v=0
    o=3cxPS 365189660672 220217737217 IN IP4 192.168.1.10
    s=3cxPS Audio call
    c=IN IP4 192.168.1.214
    t=0 0
    m=audio 51226 RTP/AVP 0 8 9 2 3 18 4 101
    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+4rEXe7iRpl341EvnHW6E7piV217ohwCASQDYYnm
    a=fmtp:101 0-16
    a=ptime:20
    a=rtpmap:0 pcmu/8000
    a=rtpmap:8 pcma/8000
    a=rtpmap:9 g722/8000
    a=rtpmap:2 g726-32/8000
    a=rtpmap:3 gsm/8000
    a=rtpmap:18 g729/8000
    a=rtpmap:4 g723/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    a=encryption:eek:ptional

    16:59:22 SIP_SI> [GW GW_SIP_0] < Stack: New Context Needed
    16:59:22 SIP_TR> [GW] > Stack: SIP/2.0 404 Not Found
    Call-ID: ZGI1NDdmNWZiMGQzM2IyMzIxOWY2MGYzZWI5ZWQ2NGY.
    CSeq: 1 INVITE
    From: "10" <sip:10@192.168.1.10:5060>;tag=33278d71
    To: <sip:01653669000@192.168.1.4:5060>;tag=be21e000ca98174
    Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d87543-20496a4d063a4e41-1--d87543-;rport
    Content-Length: 0
    User-Agent: Patton SN4634 3BIS UI MxSF v3.2.8.45 00A0BA02E092 R4.2 2008-01-17 H323 SIP BRI


    16:59:22 SIP_TR> [GW] < Stack: ACK sip:01653669000@192.168.1.4:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d87543-20496a4d063a4e41-1--d87543-;rport
    To: <sip:01653669000@192.168.1.4:5060>;tag=be21e000ca98174
    From: "10"<sip:10@192.168.1.10:5060>;tag=33278d71
    Call-ID: ZGI1NDdmNWZiMGQzM2IyMzIxOWY2MGYzZWI5ZWQ2NGY.
    CSeq: 1 ACK
    Content-Length: 0


    16:59:22 SIP_SI> [GW GW_SIP_0] < Stack: New Context Needed
    16:59:22 SIP_TR> [GW] < Stack: INVITE sip:01653669000@192.168.1.4:5062 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d87543-7d4bf93e8145255d-1--d87543-;rport
    Max-Forwards: 70
    Contact: <sip:10@192.168.1.10:5060>
    To: <sip:01653669000@192.168.1.4:5062>
    From: "10"<sip:10@192.168.1.10:5060>;tag=6f09a506
    Call-ID: MTIzNjBjM2QxZmZhOTkwYTcxNGE1Y2UxOGQ3MWE4N2E.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
    Content-Type: application/sdp
    User-Agent: 3CXPhoneSystem 5.1.4287.0
    Content-Length: 491

    v=0
    o=3cxPS 93281320960 55801020417 IN IP4 192.168.1.10
    s=3cxPS Audio call
    c=IN IP4 192.168.1.214
    t=0 0
    m=audio 51226 RTP/AVP 0 8 9 2 3 18 4 101
    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+4rEXe7iRpl341EvnHW6E7piV217ohwCASQDYYnm
    a=fmtp:101 0-16
    a=ptime:20
    a=rtpmap:0 pcmu/8000
    a=rtpmap:8 pcma/8000
    a=rtpmap:9 g722/8000
    a=rtpmap:2 g726-32/8000
    a=rtpmap:3 gsm/8000
    a=rtpmap:18 g729/8000
    a=rtpmap:4 g723/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    a=encryption:eek:ptional

    16:59:22 SIP_SI> [GW GW_SIP_1] < Stack: New Context Needed
    16:59:22 SIP_TR> [GW] > Stack: SIP/2.0 404 Not Found
    Call-ID: MTIzNjBjM2QxZmZhOTkwYTcxNGE1Y2UxOGQ3MWE4N2E.
    CSeq: 1 INVITE
    From: "10" <sip:10@192.168.1.10:5060>;tag=6f09a506
    To: <sip:01653669000@192.168.1.4:5062>;tag=c6f90b754609e46
    Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d87543-7d4bf93e8145255d-1--d87543-;rport
    Content-Length: 0
    User-Agent: Patton SN4634 3BIS UI MxSF v3.2.8.45 00A0BA02E092 R4.2 2008-01-17 H323 SIP BRI


    16:59:22 SIP_TR> [GW] < Stack: ACK sip:01653669000@192.168.1.4:5062 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d87543-7d4bf93e8145255d-1--d87543-;rport
    To: <sip:01653669000@192.168.1.4:5062>;tag=c6f90b754609e46
    From: "10"<sip:10@192.168.1.10:5060>;tag=6f09a506
    Call-ID: MTIzNjBjM2QxZmZhOTkwYTcxNGE1Y2UxOGQ3MWE4N2E.
    CSeq: 1 ACK
    Content-Length: 0


    16:59:22 SIP_SI> [GW GW_SIP_1] < Stack: New Context Needed
    logout
    Press 'yes' to logout, 'no' to cancel : yes
    Goodbye
     
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  3. mtos122

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    In Patton “/ Telephony / Call-Router / SIP Interface IF_SIP_0” there was nothing in “call-routing-destination”
    Changed to interface = ISDN and worked!
     
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  4. Banana Jack

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    Wow that's great - it works for me too!! Well done and thank you for posting your solution. I don't know why 3CX didn't configure this option in its automated config file.

    Thanks again
    Glenn
     
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