I’m having very intermittent (4 or 5 out of 60 calls) one way audio problems. I’ve done everything I can think of to resolve this and nothing has made any difference.
Asterisk: 13.5.0 and Freepbx 13.0.2, Grandstream hardphones
My Asterisk system is based on a VPS, firewall is completely open towards our SIP provider (and also our phones) so no issues there. When listening to recordings of the affected calls, I can hear both parties but it’s clear that one party (always the callee) cannot hear the caller, but the caller can hear the callee. Given that both parties are present in the recordings, it would occur to me that the issue is between our Asterisk system and our phones.
All of the phones are behind DrayTek routers, I have tried them regular (with NAT all turned to yes / keep alive, also with it all turned off), using STUN (which reports a restricted port cone nat), and now I even resorted to changing the ports so each phone is using individual ports and manually port forwarding for each phone, and I’m STILL getting one way audio, but only intermittently
Not being able to reliably test it is frustrating, but how can I be having the exact same issue between the Asterisk server and the phones despite being behind NAT, behind NAT with STUN and via port forwarding? Given that the phones are now open ports I cannot understand how this one way audio can be happening now.
Any suggestions to stop me from going completely mad would be much obliged. Previously we had our Asterisk server on the local network with the phones, and had no issues for years.