[Asterisk-Dev] SIP channels not cleared

[Jerris, Michael MI]

> 
> > [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of 
> Chee Foong
> > 
> > Hello,
> > 
> > My vendor's system (Pactolus) sends a BYE for INVITE sent 
> by Asterisk 
> > before sending OK to asterisk to terminate a call.
> > This caused SIP channels is Asterisk not being cleared up 
> (described 
> > in my previous post). I have looked at chan_sip.c, I am thinking of 
> > modify the handle_request_bye to accomodate my vendor's 
> system but I 
> > am not sure how to do it because I am not proficient in C 
> programming.
> > 
> > I am looking at handle_request_bye function. I am 
> suspecting that the 
> > user counter is not decreased. If I put a update_user_counter(p, 
> > DEC_OUT_USE) somewhere in the handle_request_bye function would it 
> > help clearing the channel?
> > 
> > I know the other end is faulty, but I still want to make this work.
> > 
> > Thanks
> > 
> > CCF
> > 
> 
> This should already be fixed in 1.2 beta 1 and cvs head.  
> Details of the fix are here:
> 
http://lists.digium.com/pipermail/asterisk-cvs/2005-August/007401.html

Clicked send to fast, sorry.

Mike