[Asterisk-Dev] SIP channels not cleared

[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


-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of Olle E.
Johansson
Sent: Sunday, August 21, 2005 18:56
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] SIP channels not cleared


Chee Foong wrote:
> But I still dont understand why asterisk response to the BYE with OK since
> it is not permited at the stage and leave the channels hanging.
>
As I reported in another mail, I just found out that the caller can send
a BYE on a non-answered call, I was wrong. We just have to cancel the
outstanding INVITE with a 487 and then answer 200 OK on the BYE. Test
this patch:

http://bugs.digium.com/view.php?id=4994

/O
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