On 24/05/11 12:08, Tony Mountifield wrote:
OK, thanks. Sounds like there was some kind of issue at the ITSP then.
I have seen this happen with broken SIP-ALGs in routers too. The ITSP
sends the BYE but for some reason a broken SIP ALG will not deliver the
packet to the right place. The ITSP will resend the BYE several times
if they don't receive the responding OK message (or some error such as
481 etc....) but after a few attempts it's pointless them continuing.
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.
SIP session timers is what you need for that. Implemented in Asterisk 1.8.
That's useful to know. Planning on moving from 1.2 to 1.8 over the next
few months.
Setting absolute timeouts on all calls might help too:
http://www.the-asterisk-book.com/unstable/funktionen-timeout.html
Although it's a bit of a balancing act, it can be used to limit these
things to a couple of hours rather than having "stuck" calls going on
for days.
cheers,
Paul.
Cheers
Tony
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