[asterisk-users] SIP per-call heartbeat?

[Tony Mountifield]

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: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org