problem with taking phone on the road

Discussion in '3CX Phone System - General' started by bblokey, Apr 29, 2008.

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  1. bblokey

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    I have taken a voip phone with me to work after some testing locally on my subnet. At home it works fine I can place calls from it and receive. But when I took the phone to work with me and hooked it up the network I am not able to place or receive calls. It is showing that the line is registered with my 3cx server and when I place a call to the server and dial the extension it finds the ip address of the phone. But it will not route the call it will just send it to VM. Here is the output of the server monitor just for that call.

    08:43:50.234 Call::Terminate [CM503008]: Call(17): Call is terminated
    08:43:50.218 Call::Terminate [CM503008]: Call(17): Call is terminated
    08:43:48.156 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:800
    08:43:38.265 CallCtrl::eek:nLegConnected [CM503007]: Call(17): Device joined: sip:
    08:43:38.265 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:800
    08:43:38.250 CallCtrl::eek:nRerouteReq [CM503005]: Call(17): Forwarding: IVR:RecordMessage@[Dev]
    08:43:32.968 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(17): Calling: Ext:102@[Dev:sip:102@192.168.0.126:5060;transport=tcp]
    08:43:29.312 MediaServerReporting::DTMFhandler [MS211000] C:17.1: 192.168.0.100:16390 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.

    Anyone have any idea to the problem? :cry:
     
  2. bblokey

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    Here is when I try to dial an extension from the phone to a softphone at home.

    08:50:55.109 Call::Terminate [CM503008]: Call(19): Call is terminated
    08:50:55.109 CallCtrl::eek:nIncomingCall [CM502001]: Source info: From: 19; To: [sip:102@AYWEB.SYTES.NET];tag=5021a9313b98fc73[sip:500@AYWEB.SYTES.NET]
    08:50:55.109 CallCtrl::eek:nIncomingCall [CM503012]: Call(19): Incoming call rejected, caller is unknown; msg=SipReq: INVITE 500@ayweb.sytes.net tid=36f3bceaf936fe3b cseq=INVITE contact=102@192.168.0.126:5060 / 6068 from(wire)
    08:50:54.296 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:500@AYWEB.SYTES.NET SIP/2.0
    Via: SIP/2.0/TCP 192.168.0.126:5060;branch=z9hG4bKeef3d304d626ca41;received=12.5.86.130;rport=35145
    Max-Forwards: 70
    Route: [sip:71.170.203.108;transport=tcp;lr]
    Contact: [sip:102@192.168.0.126:5060;transport=tcp]
    To: [sip:500@AYWEB.SYTES.NET]
    From: [sip:102@AYWEB.SYTES.NET];tag=5021a9313b98fc73
    Call-ID: 034dfb62d4281514@192.168.0.126
    CSeq: 6067 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE, PRACK, MESSAGE
    Supported: replaces, timer, path
    User-Agent: Grandstream GXP2020 1.1.5.15
    Content-Length: 0


    08:50:54.296 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE 500@ayweb.sytes.net tid=eef3d304d626ca41 cseq=INVITE contact=102@192.168.0.126:5060 / 6067 from(wire)
     
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