FreePBX Distro15
asterisk 16.2
PJSIP trunks
Chan_SIP extensions
Site to PBX VPN
I know there are dozens of posts about this issue and most come down to suggesting that the local firewall has a udp timeout or sip port translation issue.
I was following those posts then it hit me. The PBX in question is in the cloud, not behind the local sonicwall firewall. So the trunks shouldn’t be subject to the local firewall. Or would they?
The trunk leg no, the extension leg yes.
ok, most of the posts I saw pointed to the trunks as the issue.
What you’re saying is it could be the trunk OR the extensions settings?
All firewalls carriyng the bridged call will subject their own rules.
ok, waiting for the tech support rep that has firewall access to call me back…
turns out it was the UDP timeout on the local sonicwall. it was set to 30 seconds. Changed it to 160 and outbound calls now go past the 15 min mark…
The 15 min failure is one of those numbers that jump out when debugging SIP. When an INVITE is sent from the PBX to a registered endpoint, it will carry a Session-Expires header of 1800 (by default). (An INVITE sent from the client may have a different expiry, or none at all - fire up sngrep if you want to see for yourself). So the session will last a maximum of 1800 seconds or 30 min. At the half way mark, a reINVITE is sent to update the expiry time, which requires a 200 OK and an ACK. If the reinvite signaling fails, depending on the nature of the failure, the call can end immediately at the 15 minute mark, or continue the full 1800 seconds until the expiry is reached and fail then.
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