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Digital Receptionist stopped playing prompt wav

Discussion in '3CX Phone System - General' started by pbsa, May 23, 2011.

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  1. pbsa

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    I setup a digital receptionist and added a wav file as the prompt. I then called in externally and it was all working. Then after a day or so with no change to configuration it stopped working for external calls, you hear nothing, however if you press 1 or 2 etc it completes the action. If you call internally to the digital receiptions extension (805) it plays the prompt and works as expected.

    I have restarted abyss web server, I have recorded new prompts, I have restarted 3cx to no avail. I have checked the timeout setting in the config settings is 86400.

    Can anyone give me a fix to this, or point me in the right direction.

    Logs are below which shows the call joined the digital receceptionist which has an extension of 805.

    Appreciating any help. :D

    Logs

    14:18:15.039 Currently active calls - 1: [25]
    14:17:56.670 Session 4079 of leg C:25.1 is confirmed
    14:17:56.570 [CM503007]: Call(25): Device joined: sip:805@127.0.0.1:40600;rinstance=ec594ff8dce765a6
    14:17:56.558 [CM503007]: Call(25): Device joined: sip:959262@sip03.mytel.net.au:5060
    14:17:56.548 [MS210001] C:25.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40616(40617)
    14:17:56.543 Remote SDP is set for legC:25.2
    14:17:56.534 [CM505001]: Ext.805: 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:805@127.0.0.1:5060]
    14:17:56.534 [CM503002]: Call(25): Alerting sip:805@127.0.0.1:40600;rinstance=ec594ff8dce765a6
    14:17:56.382 [CM503025]: Call(25): Calling Ext:Ext.805@[Dev:sip:805@127.0.0.1:40600;rinstance=ec594ff8dce765a6]
    14:17:56.379 [MS210002] C:25.2:Offer provided. Connection(transcoding mode): 127.0.0.1:7092(7093)
    14:17:56.354 [CM503004]: Call(25): Route 1: Ext:Ext.805@[Dev:sip:805@127.0.0.1:40600;rinstance=ec594ff8dce765a6]
    14:17:56.352 [CM503010]: Making route(s) to <sip:805@192.168.1.4:5060>
    14:17:56.342 [CM505003]: Provider:[mytel] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] PBX contact: [sip:959262@203.208.66.244:5060]
    14:17:56.333 [CM503001]: Call(25): Incoming call from 0390147788@(Ln.10000@mytel) to <sip:805@192.168.1.4:5060>
    14:17:56.326 [MS003005] C:25.1: Failed to create Endpoint: (destination=203.166.103.242)EndPoint: ID=0000003D@127.0.0.1:5482(EXTERNAL) LOGID=C:25.1 Status: MSEP_FAILED RTP:203.208.66.244:9028 RTCP:203.208.66.244:9029 STUN RTP:0.0.0.0:0 STUN RTCP:0.0.0.0:0 Coder: NOT SET 101:telephony-event Party ptime:20 Party RTP:0.0.0.0:0 Party RTCP:0.0.0.0:0 Decoders: <empty>
    14:17:56.325 [MS003003] C:25.1: RTCP socket 203.208.66.244:9029 binding to 192.168.1.41:9029 failed with error code 10049
    14:17:56.325 [MS003002] C:25.1: RTP socket 203.208.66.244:9028 binding to 192.168.1.41:9028 failed with error code 10049
    14:17:56.312 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:805
    14:17:56.312 Looking for inbound target: called=0390015865; caller=039014778814:17:56.307 [CM500002]: Info on incoming INVITE:
     
  2. davidbenwell

    davidbenwell Active Member

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    this is your issue

    14:17:56.326 [MS003005] C:25.1: Failed to create Endpoint: (destination=203.166.103.242)EndPoint: ID=0000003D@127.0.0.1:5482(EXTERNAL) LOGID=C:25.1 Status: MSEP_FAILED RTP:203.208.66.244:9028 RTCP:203.208.66.244:9029 STUN RTP:0.0.0.0:0 STUN RTCP:0.0.0.0:0 Coder: NOT SET 101:telephony-event Party ptime:20 Party RTP:0.0.0.0:0 Party RTCP:0.0.0.0:0 Decoders: <empty>
    14:17:56.325 [MS003003] C:25.1: RTCP socket 203.208.66.244:9029 binding to 192.168.1.41:9029 failed with error code 10049
    14:17:56.325 [MS003002] C:25.1: RTP socket 203.208.66.244:9028 binding to 192.168.1.41:9028 failed with error code 10049

    Can you run the 3CX firewall checker and let us know the output?

    also you said that if you press one of the options then it works. once the call is connected do you have audio both ways?

    it may be that RTP audio is not being sent / received
     
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