[asterisk-users] SIP call interrupted after 64 seconds

[Olle E Johansson]

17 dec 2007 kl. 18.49 skrev Roger Schreiter:

> Hi,
>
> some months ago, I had the problem with an asterisk-1.4.x-
> Version, that some calls (but not all) were interrupted
> 64 seconds after connect (a call limit of 86400 seconds
> was installed using the S()-parameter).
>
> It was just a test machine, and later, I switched to callweaver,
> and the problem had gone. Thus, I never investigated this problem.
>
> Now, I upgraded a machine for production use to asterisk-1.4.8,
> and do encounter the same problem.
>
> I have other asterisk machines running, using the same
> dialplan, without this problem.
>
> Did anyone else observe this strange behaviour of calls ending
> after 64 secondes of uptime?

There is a hidden reason somewhere and you need to add
verbose logging to your Asterisk, maybe also debug logging
so that you can find out what's going on - where the call fails.

With the log files, it's often very simple for a trained eye to
spot what goes on. It seems like some kind of signalling
problem as it is kind of close to the SIP timeouts.

If you think it is a bug, don't hesitate to file a bug report
and add your log output with verbose set to 4 and debug
set to 4, sip debug also turned on!

/Olle