[Asterisk-Dev] SIP channels not cleared

John Todd jtodd at loligo.com
Thu Aug 18 22:51:13 MST 2005


At 7:06 AM +0200 on 8/19/05, Olle E. Johansson wrote:
>Chee Foong wrote:
>>  OK, I under stand.
>>  So, can this be considered a bug in asterisk?
>>  Since it knows how to response to a BYE, it should also know it's time to
>>  clear the channel.
>
>The real fault here is that the other end issues a BYE when we have no
>session set up by
>INVITE/200 OK/ACK - to cancel a pending INVITE you use CANCEL, not BYE.
>That is a bug, please ask your vendor to look up CANCEL in the SIP rfc.
>
>And yes, we should be able to handle faulty devices better, but will
>concentrate our energy on being able to improve the way we handle
>devices that actually support basic SIP according to the standard. ;-)
>
>/Olle


This problem could perhaps could be resolved by implementation of 
session-timers on the Asterisk side, assuming that the UAC also 
supported (or at least did not crash on) such timers.

http://www.faqs.org/rfcs/rfc4028.html

If Asterisk sent re-INVITEs after the Session-Expires: duration, then 
it (Asterisk) could close channels which did not respond.  I would 
think that this would be something that could be set on a per-peer 
basis or globally.

I believe my previous tests with Asterisk showed that Asterisk 
supported Session-Expires: in a non-harmful way (i.e.: did not crash) 
but Asterisk did not seem to have any "hooks" for generating a 
Session-Expires: header or creation of timers.  Does anyone have any 
alternate information?  It's been a year or so since I experimented 
with equipment using session-timers.

JT



More information about the asterisk-dev mailing list