VERY long term freepbx user here. Setting up a new site a week back we went with V16 and found a LOT of stuff was broken. The system wasnt stable and didnt behave as expected. We lost all network connectivity overnight a few times and we were having weird IAX issues. So, Long story short. We rolled back to V15 and all is well, mostly The weird IAX issue persists…
We have Site A, a long established site, everything works here except the phone for their roaming gnome, and that’s a Grandstream Issue. This site has about 20 phones, 4 sip trunks to VOIPfone and is sat behind a PF box on gigabit fibre.
Site B is in the middle of no-where. Fibre is promised but the best install date is “before xmas” so they are sat on a LEO satellite system (Starlink).
The two sites are linked by an IAX2 trunk thats encapsulated in an uncmpressed VPN. A is 192.168.10.3, B is 192.168.11.3 and access to all services between networks works perfectly.
So as it stands, a call somes into site A, goes through ivr, setcid, ring groups etc everything as it should, rings the phones on both sites, anyone can answer, everything works except the truckload of jitter on Site B but it’s liveable.
Site A can call all of Site B’s phones, this all works, and Site A can transfer calls. In short, going A-=>B works perfectly
Going B to A or the trunks, it goes wrong. There’s no audio, and more than that there is no sign of anything audio stream like in the traffic. I can see IAX control data but no stream. Moreover looking at either ends at the phone/trunk level there is also no sign of rstp streams. The call progresses as it should, the call is connected and bar the lack of any audio in any direction its working. There are no errors I can see anywhere in the console or logs, codecs all match along the path. Destination/Source phone model is not a factor (They have GS, Yealink, Aastra, Poly and a Cisco on my desk which all behave the same.
It feels like there is a negotiation issue setting up the audio streams to the target phone/trunk.
A second PJSIP channel going from B to A fixes it, albeit with no jitter buffer, so it’s choppy as hell. So the phones on either end do seem to work.
Adding a system recording to misc destintions on Site A works and gives me the recording on both networks, so initiation doesnt seem to be the issue, although I dont understand why I get that audio stream just fine.
I’ve checked codecs, traffic inside and outside the tunnel, briefly tied to punch IAX through Starlink’s CG-NAT (Didn’t work) and I’m now out of ideas. I’m going to attempt to route through our server farm later and see it it really is a site A issue but I just have run out of ideas.