Conference Module One way Audio

Have a weird issue here, but I can recreate it consistently.

Running Asterisk 13 and FPX 13 when a person calls in from the outside into the conference bridge with the Name announce feature enabled. They will be prompted to record their name like you would expect. They hit “#” and then they will no longer receive any further audio on the bridge. An internal user can hear them but the external user can not hear others.

If the bridge is accessed completely internally the issue does not occur. If you disable the name announce feature the bridge operates correctly for both internal and external users.

If you can do it economically, could you send the output from /var/log/asterisk/full around where this is happening. We don’t need everything in the file - just the 30-ish lines where the problem appears to be happening.

I’m interested specifically in the part around where the system plays the recording to record their name and after. If you want to do what I’m going to do - look for error messages.

My theory is that the file is not getting properly transcoded to the codec the call is using, and when the file gets replayed, the codec gets switched to something that isn’t supported by the external call. My experience with a similar problem is from when Chan-SCCP-B comes off “on hold” and the music is a different codec than what the phone negotiated. It causes dead-air just like you are describing.

The more common possibility is that, when the conf-bridge starts (and connects your external calls to the conference), it is getting assigned to an RTP port that you are not passing through your firewall. Check your firewall settings to make sure you are allowing all of the UDP range associated with your PBX and that they are properly redirected to the PBX.