[asterisk-users] Channels hanging when SIP phone gets resetduring call

[Steve Langstaff]

Are the RTP timers applicable with canreinvite=yes ? 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Olle E Johansson
> Sent: 22 February 2007 10:49
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Channels hanging when SIP phone 
> gets resetduring call
> 
> 
> 21 feb 2007 kl. 12.54 skrev Steve Langstaff:
> 
> > Hi All.
> >
> > This is on Asterisk 1.2.13
> >
> > I place a call between 2 SIP phones (with canreinvite=yes, 
> > qualify=yes).
> >
> > I reset the phones (so they don't have time to say BYE).
> >
> > Asterisk seems to think that the call is still ongoing. 
> This persists 
> > until I do a 'restart now'.
> Check the RTP timers in sip.conf. They will hangup the call 
> if there's no audio.
> 
> /O
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