[asterisk-users] How to automaticaly close calls whenAsterisk didn't receive the bye request ?

[Raj Jain]

The rtptimeout feature has a few limitations:

. It is ineffective when the RTP is not terminated on Asterisk.

. It can cause false session hangups if the remote SIP UA does not support
silence suppression

. The companion rtpholdtimeout can cause false hangups if you make incorrect
judgment on how long a call hold can last.

. The rtptimeout period is not negotiated throughout the SIP signaling path
i.e. between the UAC, UAS, and intermediary proxies. So it does not help
clear the session state throughout the network (when your BYE doesn't make
it to all the entities in the SIP signaling path).

The SIP session-timers feature addresses all of the above limitations.

--
Raj



Jared,
> I would think of using rtptimeout. There is a reason why you did not
> mention
> it and I am curious as to why.
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080103/037e7d00/attachment.htm