One of our customers has an Asterisk conference bridge connected to a SIP trunk from an ITSP. Yesterday, they had two inbound calls that didn't get hung up properly. From the tcpdump SIP trace that we have running continuously, I can see that no BYE was received by the bridge, and when some hours later the hangup was forced from the bridge end, the bridge sent a BYE to which it received a 481 Call Leg/Transaction Does Not Exist.
Since SIP is UDP, this situation must occur from time to time, and I wondered if it is possible to configure any kind of per-call SIP heartbeat so that a dead call could automatically be identified with a 481 response much sooner. Cheers Tony -- Tony Mountifield Work: [email protected] - http://www.softins.co.uk Play: [email protected] - http://tony.mountifield.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
