[SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint

[Brett Woollum]

Hello, 

We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. Everything seems to be working well, until today. 

I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. 

If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. 

What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. 

Thanks! 
Brett 
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