I’m running Asterisk 11.7.0 on FreePBX version 18.104.22.168. I’ve tested this on newer and older versions of Asterisk/FreePBX with the same results. I’ve tried this with both hosted and premise based boxes. Each tested PBX has its own dedicated static, routable IP that is NATed to an internal IP. Ports 5060 and 10000-20000 are open to the world.
The issue is that when we call an external number that is hosted on the same PBX as the number that is being called, we receive the ringing on the called phone, but there is no audio when the call is answered in either direction. This was not a problem until recently when our SIP provider made a change to the way they handle routing.
Previously, if a call was made it would be passed out of the trunk to our SIP provider, the provider would set up the call to a media gateway, then that media gateway would pass the traffic back to the PBX. Now, the provider recognizes that the DID is hosted on their system and passes the traffic back to the originating IP, completely circumventing the media gateway (and the associated cost).
I’ve looked at our firewall, as I assumed this was a NAT no audio issue, but all of the rules appear to be correct and I’m not seeing anything being blocked in the firewall logs.
Trunk settings are here:
Any advice is greatly appreciated.