Need help with pjsip trunks connecting to voip innovations

This has to be a timer issue but I can’t find it. Calls end after about 30 seconds and it’s the PBX doing it. The freepbx and the VI wiki’s dont mention any settings beyond what IPs to match/permit.

PBX Version:
PBX Distro: 12.7.8-2107-3.sng7
Asterisk Version: 16.20.0

Hoping this is something simple. Here are the trunk settings

In the log i can see it go through all the checks, then the phone rings, then it’s answered, and 32 seconds later it hangs up for whatever reason

[2021-09-22 10:46:18] VERBOSE[3994][C-0000000c] app_dial.c: PJSIP/9998-00000007 is ringing
[2021-09-22 10:46:20] VERBOSE[3994][C-0000000c] app_dial.c: PJSIP/9998-00000007 answered PJSIP/VI_In_PJSIP_1-00000006
[2021-09-22 10:46:20] VERBOSE[4011][C-0000000c] bridge_channel.c: Channel PJSIP/9998-00000007 joined ‘simple_bridge’ basic-bridge
[2021-09-22 10:46:20] VERBOSE[3994][C-0000000c] bridge_channel.c: Channel PJSIP/VI_In_PJSIP_1-00000006 joined ‘simple_bridge’ basic-bridge
[2021-09-22 10:46:28] WARNING[31017] chan_sip.c : Timeout on dd787779d157786183e61fbf9ffe0b28 on non-critical invite transaction.
[2021-09-22 10:46:52] VERBOSE[3994][C-0000000c] bridge_channel.c: Channel PJSIP/VI_In_PJSIP_1-00000006 left ‘simple_bridge’ basic-bridge

In Asterisk SIP settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config you must also restart Asterisk.
In the pjsip specific settings, External IP Address and Local Network should normally be left blank.

If no luck, at the Asterisk command prompt, type
pjsip set logger on
make a failing incoming call, paste the Asterisk log for the call at and post the link here. Also, if this is an on-site PBX, post details about your router/firewall and any other intermediate network equipment.

It was the local network. Not sure how that was missed. Thanks.

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