Issue: inbound calls to the SIP Trunk are, intermittently, not coming through to the Asterisk server. I believe due to some weird trunk state.
When using ‘sip set debug on’ I notice the SIP Trunk does a REGISTER to the VoIP provider and getting a 200 OK. At this state, the Asterisk server properly receives inbound calls.
A couple of seconds later, Asterisk does an OPTIONS to the VoIP provider. It receives a 401 Unauthorized. At THIS state, the Asterisk server CAN NOT receive inbound calls.
The Asterisk server cycles between these states: REGISTER + OK, then OPTIONS + Unauthorized. In order for the Asterisk to see an inbound call from the VoIP provider, it needs to be in the REGISTER + OK state.
Is there some configuration missing? The server is non-NAT.
I also notice a couple of seconds after the SIP Trunk has registered that Asterisk debug is telling me it is “Scheduling destruction of SIP dialog X”, where X is the trunk REGISTER dialog. Perhaps this is normal behavior?