hi everyone,
I have set up a freebpx as a vm with a pjsip trunk using an internal and an external network.
PBX Version:15.0.17.43
PBX Distro:12.7.8-2107-3.sng7
Asterisk Version:16.19.0
When I use my personal UCP phone widget (connecting to external IP) to call my external mobile the SIP signalling works but there is no audio in both directions.
Using sngrep on the freepbx it seems like the SDP information is correct, showing IPs and Ports of my pbx and my provider.
But when I enable rtp debugging on the asterisk console I see that rtp packets are sent from the pbx with the local router interface as destination on a port that is out of range for 10000 - 20000. The Firewall passes UDP for port range 10000 - 20000
As soon as I pick up my mobile phone the rtp stops completely.
I would expect that the rtp destination address is the provider IP with a port between 10000 and 20000 as described in the SDP bodies.
The logs show me that strict RTP learns IP and port from my websocket connection. That most definitely is not what I want to achieve.
I disabled direct dial for my extension. Should I disable strict RTP? Can anyone help me on this please?
Thank you!