[asterisk-dev] Bug in chan_sip

[Venefax]

Will the fix be applied to 1.4 as well?

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Jon Bonilla (Manwe)
Sent: Wednesday, December 03, 2008 5:18 AM
To: asterisk-dev at lists.digium.com
Subject: Re: [asterisk-dev] Bug in chan_sip

El Wed, 3 Dec 2008 09:14:44 +0100
"Saúl Ibarra" <saghul at gmail.com> escribió:

> Manwe is very right, this could break the communication between
> Asterisk and a SIP proxy. By reading the commints list, I see this bug
> has been introduced in revision 158071
> (http://svn.digium.com/view/asterisk?view=rev&revision=158071).
> 
> We should destroy the dialog only when a 481 or 408 is received, and
> treat other responses to BYE requests as usual.
> 
> 
> 

This bug is also in version 1.6.1beta3:

line 16069 of chan_sip.c

        if (resp >= 400 && resp < 500 && sipmethod == SIP_BYE) {
                p->needdestroy = 1;
                return;
        }





-- 
First they ignore you.
Then they laugh at you.
Then they fight you.
**Then you win.**

DaHjaj jaj QaQ Daghajjaj :)

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev