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No RTP packets were received

Discussion in '3CX Phone System - General' started by milangarg, Aug 18, 2011.

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  1. milangarg

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    Hi,

    I am trying to call voxeo prophecy through 3cx .
    The SIP connection goes through
    But the call automatically disconnects.
    I suspect that 3cx phone system is disconnecting the call while waiitng for RTP packets.
    the contents in the log are
    12:32:57.850 [CM503008]: Call(3): Call is terminated
    12:32:55.676 [MS105000] C:3.2: No RTP packets were received:remoteAddr=192.168.1.5:10010,extAddr=0.0.0.0:0,localAddr=192.168.1.5:7014
    12:32:54.167 [CM503007]: Call(3): Device joined: sip:10005@192.168.1.5:5071;transport=udp
    12:32:54.166 [CM503007]: Call(3): Device joined: sip:10001@192.168.1.5:61496;transport=udp
    12:32:54.160 [CM505001]: Ext.10005: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [VCS] PBX contact: [sip:10005@192.168.1.5:5060]
    12:32:54.160 [CM503002]: Call(3): Alerting sip:10005@192.168.1.5:5071;transport=udp
    12:32:54.010 [CM503025]: Call(3): Calling Ext:Ext.10005@[Dev:sip:10005@192.168.1.5:5071;transport=udp]
    12:32:54.003 [CM503025]: Call(3): Calling Ext:Ext.10005@[Dev:sip:10005@192.168.1.5:5071;transport=tcp]
    12:32:53.961 [CM503004]: Call(3): Route 1: Ext:Ext.10005@[Dev:sip:10005@192.168.1.5:5071;transport=tcp,Dev:sip:10005@192.168.1.5:5071;transport=udp]
    12:32:53.960 [CM503010]: Making route(s) to <sip:10005@192.168.1.5>
    12:32:53.956 [CM505001]: Ext.10001: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [IM-client/OMA1.0 IMSDroid/v1.0.320 (doubango r501)] PBX contact: [sip:10001@192.168.1.5:5060]
    12:32:53.954 [CM503001]: Call(3): Incoming call from Ext.10001 to <sip:10005@192.168.1.5>

    although I see that the RTP ports are mapped properly as shown in the sip call headers.
    It seems that the call is timing out.

    Can I somehow prevent this time out ?

    Regards
     
  2. willow

    willow Member

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    can we see the sip header? also run the 3cx firewall and make sure the rtp ports are not blocked.
     
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  3. milangarg

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    Hi,

    I have uploaded the logs that were generated on the voxeo prophecy side and contain the SIP calls.
    In 3cx I have disabled the STUN server and given my machines STATIC IP .The IP that the router has assigned to my machine.
    Moreover all my components i.e. call originating sip phone,3cx pbx and voxeo prophecy are on the same machine.
    I ran the firewall checker and it was giving me error 51.

    I am still not clear whether the SIP calls are actually getting routed through the firewall when all the componenets are on the same local machine.


    Regards
     
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