[SR-Users] Kamailio - Asterisk: Handling loss of SIP BYE and dangling channels

Hi, I'm wondering what the best approach to handling a SIP dialog when one endpoint disappears/fails to send the BYE message. I have Kamailio as a proxy for all mobile (iPhone/Android) SIP clients. Occasionally, the user hangs up the call but no BYE message is received. This means that Asterisk has an open channel even though there is no client. Kamailio also continues to receive successful registrations from the SIP client so the endpoint is not down completely. Is Kamailio the appropriate place to handle this situation? What do you recommend? If not could you point me in the right direction? RTP timeout? Asterisk? The SIP client itself? Thanks for your help. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us