I’m just getting started with FreePBX and am having an issue with RTP port ranges. I am operating behind a NAT firewall. I have specified the RTP port range in FreePBX to be 10001 to 20000 and mapped those same ports on UDP on the firewall to open those ports to the internet. When connecting with a SIP client or trying to connect SIP trunks, my router is showing connections on ports outside of this range. This causes there to be no audio in the call. If I then change the range and open the ports that I see it’s trying to connect to, everything works again (eg: I observe my trunk provider flowroute connecting in the 32XXX range, so I change the RTP ports to be 32000-34000 and it starts working).
I feel like I must be missing a setting - shouldn’t FreePBX negotiate the allowable RTP port range with the client so that it knows what ports to use? Or might I just need to open all UDP ports to RTP?