Ok we are stuck. I don’t think it’s a good idea to go ahead and try finding out which services on the freepbx use what port and what means of communication and set up rules to make sure all of them get routed the correct way.
If this system has to be running reliably for a client I don’t want to have to go out and fix it again if some update changes the port something uses to connect to the internet or if for example somebody installs a new module and this all of a sudden wants to connect on a port that was excluded.
Also I just don’t see how it is a wise idea to pretty much route the whole udp port range to the SIP providers network. (The routed udp range would need to be from 1024 to 65535 because that’s all the ports which can be used by rtp and i don’t have one clue what my SIP provider has set the range to.)
And even if I go and set up custom rules for everything that has to connect to the internet, if it’s within the range that the rtp uses if the rtp chooses to use this port it will again land in the wrong place and the calls will be with no audio.
My initial idea was to be able to discuss why my routing setup which I posted at the beginning of this thread was in any way faulty or not working but I guess I’ll now have to go to a linux networking forum with that.