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Digital Receptionist - Transfer Failed

Discussion in '3CX Phone System - General' started by SnAzBaZ, Jun 14, 2013.

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  1. SnAzBaZ

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    Hello,

    We are using 3CX 9. I have a digital receptionist set up to transfer calls to various ring groups. For the last two years it has been working just fine, today it's doing something strange. Instead of transferring the call to 601 (Our ring group) it transfers to 111. Instead of transferring to 600 (Operator Ring Group) it transfers to 000.

    As 111 and 000 do not exist as extensions I get the "Call Transfer Failed" message and then the receptionist starts again.

    Here is a grab from the log file for a call where this occurs. Note phone numbers and IP's have been manually changed.

    Code:
    08:56:39.402|.\CallLeg.cpp(124)|Log5||CallLeg::onNewCall:[CM500002]: Info on incoming INVITE:
    INVITE sip:xxx@3cx.X-X.co.uk SIP/2.0
    Via: SIP/2.0/UDP x:5060;branch=b8b0b1d3679010be
    Max-Forwards: 69
    Contact: "07709 xxx"<sip:xxx@81.x.30.x:5060>
    To: "X Number Block"<sip:xxx@3cx.X-X.co.uk>
    From: "07709 xxx"<sip:xxx@sip.x.co.uk>;tag=b8b0b1d3679010be
    Call-ID: b8b0b1d3679010be@sip.x.co.uk
    CSeq: 2379 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, MESSAGE
    Supported: replaces, callerid
    User-Agent: AASIP
    Content-Length: 0
    Remote-Party-ID: "07709 xxx" <sip:xxx@sip.x.co.uk>;privacy=off;screen=yes
    
    <br>
    08:56:39.417|.\Line.cpp(301)|Log5||LineCfg::getInboundTarget:Looking for inbound target: called=xxx; caller=xxx<br>
    08:56:39.417|.\Line.cpp(353)|Log2||LineCfg::getInboundTarget:[CM503012]: Inbound office hours rule (unnamed) for 10000 forwards to DN:621<br>
    08:56:39.417|.\CallCtrl.cpp(210)|Log2||CallCtrl::onIncomingCall:[CM503001]: Call(8): Incoming call from xxx@(Ln.10000@xxx) to <sip:621@192.168.5.6:5060><br>
    08:56:39.433|.\Line.cpp(1336)|Log2||Line::printEndpointInfo:[CM505003]: Provider:[xxx] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [AASIP] PBX contact: [sip:xxx@81.x.11.x:5060]<br>
    08:56:39.433|.\CallLeg.cpp(1095)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:8.1<br>
    08:56:39.433|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:8.1:Offer received. RTP connection: 81.187.30.111:16577(16578)<br>
    08:56:39.433|.\CallCtrl.cpp(398)|Log3||CallCtrl::onSelectRouteReq:[CM503010]: Making route(s) to <sip:621@192.168.5.6:5060><br>
    08:56:39.433|.\CallCtrl.cpp(530)|Log2||CallCtrl::onSelectRouteReq:[CM503004]: Call(8): Route 1: Ext:Ext.621@[Dev:sip:621@127.0.0.1:40600;rinstance=1b798b3967d286a1]<br>
    08:56:39.464|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210002] C:8.2:Offer provided. Connection(transcoding mode): 127.0.0.1:7004(7005)<br>
    08:56:39.464|.\Target.cpp(430)|Log2||Target::makeOneInvite:[CM503025]: Call(8): Calling Ext:Ext.621@[Dev:sip:621@127.0.0.1:40600;rinstance=1b798b3967d286a1]<br>
    08:56:39.636|.\CallLeg.cpp(289)|Log3||CallLeg::onAnswer:[CM503002]: Call(8): Alerting sip:621@127.0.0.1:40600;rinstance=1b798b3967d286a1<br>
    08:56:39.636|.\Extension.cpp(1153)|Log3||Extension::printEndpointInfo:[CM505001]: Ext.621: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:621@127.0.0.1:5060]<br>
    08:56:39.636|.\CallLeg.cpp(1095)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:8.2<br>
    08:56:39.636|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210001] C:8.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40610(40611)<br>
    08:56:39.636|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210003] C:8.1:Answer provided. Connection(transcoding mode[unsecure]):81.187.11.250:9112(9113)<br>
    08:56:39.652|.\CallCtrl.cpp(699)|Log2||CallCtrl::onLegConnected:[CM503007]: Call(8): Device joined: sip:xxx@xxx.call.me.uk:5060<br>
    08:56:39.652|.\CallCtrl.cpp(699)|Log2||CallCtrl::onLegConnected:[CM503007]: Call(8): Device joined: sip:621@127.0.0.1:40600;rinstance=1b798b3967d286a1<br>
    08:56:39.792|.\CallLeg.cpp(550)|Log5||CallLeg::onConfirmed:Session 169 of leg C:8.1 is confirmed<br>
    08:56:50.636|.\SLServer.cpp(830)|Log2|MediaServer|MediaServerReporting::DTMFhandler:[MS211000] C:8.1: 81.187.30.111:16577 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.<br>
    08:56:54.167|.\CallLeg.cpp(1095)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:8.2<br>
    08:56:54.167|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:8.2:Offer received. RTP connection: 127.0.0.1:40610(40611)<br>
    08:56:54.167|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210003] C:8.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7004(7005)<br>
    08:56:54.386|.\CallLeg.cpp(561)|Log5||CallLeg::onRefer:Refer: from=<sip:621@127.0.0.1:5060>;tag=b621d507; to="07709 xxx"<sip:07709264430@127.0.0.1:5060>;tag=9f239858; RefTo=<sip:111@127.0.0.1:5060><br>
    08:56:54.386|.\Line.cpp(1336)|Log2||Line::printEndpointInfo:[CM505003]: Provider:[xxx] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [AASIP] PBX contact: [sip:xxx@81.x.11.x:5060]<br>
    08:56:54.386|.\CallCtrl.cpp(398)|Log3||CallCtrl::onSelectRouteReq:[CM503010]: Making route(s) to <sip:111@127.0.0.1:5060><br>
    08:56:54.386|.\CallCtrl.cpp(474)|Log2||CallCtrl::onSelectRouteReq:[CM503014]: Call(8): No known route to target: <sip:111@127.0.0.1:5060><br>
    08:56:54.386|.\Call.cpp(977)|Log2||Call::RouteFailed:[CM503016]: Call(8): Attempt to reach <sip:111@127.0.0.1:5060> failed. Reason: Not Found<br>
    08:56:54.605|.\CallLeg.cpp(1095)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:8.2<br>
    08:56:54.605|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:8.2:Offer received. RTP connection: 127.0.0.1:40610(40611)<br>
    08:56:54.605|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210003] C:8.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7004(7005)<br>
    08:56:57.480|.\Call.cpp(1226)|Log2||Call::Terminate:[CM503008]: Call(8): Call is terminated<br>
    
    Any suggestions?

    Thanks
     
  2. lneblett

    lneblett Well-Known Member

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    Just because the extensions do not exists does not mean that one cannot input the number when the digital receptionist answers.

    You might also check to ensure that the DTMF tones are being translated correctly. Call the desired DR extension directly using an internal phone and then mimic the routing that you want. If the call is handled correctly, then try the same from an external phone such as a cell and a landline or two. If they do not work as desired, you might try restarting the system and try again.
     
  3. SnAzBaZ

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    Thanks. We will look into the DTMF tones. The receptionist config is correct, the ring groups work when calling them from an internal extension. The Receptionist works when calling from an internal extension as well, but not from an external phone. We had already rebooted the server and also restarted 3CX before I posted this.
     
  4. lneblett

    lneblett Well-Known Member

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    The key thought is to do externally from more than one phone and hopefully from more than one carrier.
     
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