If I’m on the local network, calls work fine, even though it’s looping back through the modem’s IP (confirmed this by running a traceroute)
However, if I am outside the office, the call rings and I can answer; however, there is no audio, and after 30 seconds or so it automatically hangs up.
i have check sngrep and ss and don’t see any ports in the 10k to 20k range
the asterisk full log shows:
[2026-02-25 09:56:50] VERBOSE[235346][C-00000007] app_dial.c: PJSIP/101-0000000d is ringing
[2026-02-25 09:56:53] VERBOSE[235346][C-00000007] app_dial.c: PJSIP/101-0000000d answered PJSIP/102-0000000c
[2026-02-25 09:56:53] VERBOSE[235347][C-00000007] bridge_channel.c: Channel PJSIP/101-0000000d joined 'simple_bridge' basic-bridge <b9891d09-cdd2-4e92-bd7b-ccd1a2311c95>
[2026-02-25 09:56:53] VERBOSE[235346][C-00000007] bridge_channel.c: Channel PJSIP/102-0000000c joined 'simple_bridge' basic-bridge <b9891d09-cdd2-4e92-bd7b-ccd1a2311c95>
[2026-02-25 09:57:23] NOTICE[1779] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/102-0000000c' for lack of audio RTP activity in 34 seconds
i have googled and serched for a answer but have not found anything that worked. checked ports are forwarded to correct ports and ip address. external address and local networks are put into asterisk sip settings. disabled firewall and fail2ban to see if that was blocking anything.
i have a dns entry that automatiicaly updates my dynamic ip address.
to be fair this is the furthest i have gotten with freepbx. i tried a docker before and that didn’t work. i’m now running on a vm with a static ip with debian 12.