Digital Receptionist stopped playing prompt wav

[pbsa]

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.

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:[email protected]:40600;rinstance=ec594ff8dce765a6
14:17:56.558 [CM503007]: Call(25): Device joined: sip:[email protected]: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:[email protected]:5060]
14:17:56.534 [CM503002]: Call(25): Alerting sip:[email protected]:40600;rinstance=ec594ff8dce765a6
14:17:56.382 [CM503025]: Call(25): Calling Ext:Ext.805@[Dev:sip:[email protected]: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:[email protected]:40600;rinstance=ec594ff8dce765a6]
14:17:56.352 [CM503010]: Making route(s) to <sip:[email protected]: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:[email protected]:5060]
14:17:56.333 [CM503001]: Call(25): Incoming call from 0390147788@(Ln.10000@mytel) to <sip:[email protected]:5060>
14:17:56.326 [MS003005] C:25.1: Failed to create Endpoint: (destination=203.166.103.242)EndPoint: [email protected]: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:

 

[davidbenwell]

this is your issue

14:17:56.326 [MS003005] C:25.1: Failed to create Endpoint: (destination=203.166.103.242)EndPoint: [email protected]: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