I have a static IP with all proper ports forwarded. Other extensions work just fine. I have one 3cx remote worker who can hear me, but i can’t hear him when i call him. If he starts the call, all is fine. Where should I be looking?
If the remote worker is connected by VPN, you need to add his subnet to Local Networks (in Asterisk SIP Settings).
Otherwise, I suspect a buggy SIP ALG on one end or the other. If the routers have settings for SIP ALG, try turning them off. If no luck, use SIP debug (or tcpdump) to confirm that the INVITE sent by Asterisk to the extension has the correct media address and port in the SDP. Use Wireshark on the worker’s PC to see the SDP received and where he’s sending the RTP to. If that’s correct but his RTP is not arriving back at your Asterisk, you may need to capture traffic on the WAN side of your or his router.
If his (or your) modem is actually configured as a router, be careful to take that into account.
Just thought I’d answer this completely for anyone else who comes upon the issue. Allow Reload needs to be turned on in SIP Settings under CHAN_PJSIP.
(Sorry for bumping this topic but it needs to have a final answer)