I’m having an odd problem. I’ve previously had one way audio trouble with calls through my remote SIP trunk due to a crud router which wasn’t properly capable of port forwarding.
I am now behind a pfsense firewall instead which is much better, and following the guide on their own website I have SIP calls to the phone network working nicely with two way audio.
However I have now noticed there is another problem, and that is that I cannot hang up. This only seems to happen if the other party does not answer the call. The asterisk console shows the following:
Executing [phone number removed@internal:1] Set("SIP/201-09785468", "CALLERID(num)=sip ID removed") in new stack Executing [phone number removed@internal:2] Dial("SIP/201-09785468", "SIP/phone number removed@sipgate|25|trg") in new stack Called phone number removed@sipgate SIP/sipgate-0978f250 is making progress passing it to SIP/201-09785468 Spawn extension (internal, phone number removed, 2) exited non-zero on 'SIP/201-09785468' [Jun 2 22:21:49] WARNING: chan_sip.c:12984 handle_response: Remote host can't match request CANCEL to call 'email@example.com'. Giving up.
The SIP trunk I am using is sipgate basic. I have no idea what the great big long string of characters before the final @sipgate.co.uk is all about.
Is this a network configuration issue, an asterisk configuration issue, or a sipgate issue?