Ways to monitor SIP termination on an asterisk server

Solution 1:

I have come across the same problem and it's not just limited to asterisk. In the end we came up with something that worked well for us. We called it a SIP Loopback.

Basically we signed up for an ALT sip provider (flowroute.com) and setup a script that calls out via primary SIP provider to the phone number setup with our ALT provider three times an hour. The incoming phone call would check CALLERID and then if was from primary phone number post to nagios a succes message. If no succes messages were posted to nagios at least once ever hour it would alarm. You would need to write the scripts yourself as I no longer have access to them myself but it should not be hard.