[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: