[Asterisk-Users] Sip call hang up

[Eric Wieling]

The default should be "no".  Both options listen to the audio stream. 
busydetect tries to determine if it hears a busy signal and if so
disconnects the call.  callprogress tries to determine if the call has
been disconnected and disconnects the other legs of the call.  Both
options are buggy cause false hangups.

On Wed, 2003-10-15 at 13:29, Eduardo Goncalves wrote:
> On Wed, 15 Oct 2003 11:16:03 -0500
> Eric Wieling <eric at fnords.org> wrote:
> 
> > set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf
> 
> 	Thanks for the tip. Could you explain me why these options set to yes
> may cause the hang up?
> 	At this time, I don't have these options in zapata.conf. What is the
> default?
> 
> Thanks a lot
> Eduardo
> 
> > On Wed, 2003-10-15 at 09:50, Eduardo Goncalves wrote:
> > > Hi list,
> > > 
> > > 	I have a cisco 827 with 4 fxs and an * gateway, like this:
> > > 
> > > [c827]------sip-----[asterisk]-----e&m---PSTN
> > > 
> > > 	The codec used is g711alaw over a 9Mb link.
> > > 	Some calls just hang up after some minutes of conversation.
> > > 	Cisco shows
> > > a  "DisconnectText=normal call clearing (16)" and I found nothing in
> > > asterisk logs.
> > > 	Anyone can help?
> > > 
> > > thanks
> > > Eduardo
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