i have my freepbx which has a SIP trunk to the provider. The trunk is delivered via a physical cable from the provider and plugged direct into my PBX, so there is no firewall in the path.
the issue is the SIP trunk registration is 300seconds. when the SIP trunk re registration occurs if there is an existing call, the call will terminate. iso in essence we cannot have calls that go on for more than 5 minutes.
i performed a packet capture and i can see that just after the registration event the provider is sending a “Request : OPTIONS” about 2 times of which my PBX is not responding to that request, there after the on going call terminates.
Where did you capture this? You need to enable channel driver protocol logging (e.g. pjsip set logger on) and capture the complete packets as Asterisk sees them.
However, a re-REGISTER won’t break calls in progress, so I think there is some deeper underlying cause at play here.
Also you are having BYEs ignored as well in your log extract.
If you have a direct cable to them, why are you using registration at all.
i captured this on my PBX. this service is a SIP trunk so i have to register. thats the information i got from the provider. i just gave the cable information to let you know there is no NAT in the path.
Do you know what i can check as to why my PBX is ignoring the BYEs?
The full log with protocol debugging turned on will tell you whether the BYEs reached Asterisk, and should indicate any problem with them, or provide hints that something is badly wrong with Asterisk.
Your traces don’t say which ports are in use.
That’s something imposed by the provider, not something that is fundamentally needed for a SIP trunk. Many people use FreePBX trunks with IP authentication. In any case, a failed outgoing REGISTER would stop you receiving any requests from the provider, but would still allow you to send to them.
