[Asterisk-Users] Sip call hang up

[Eduardo Goncalves]

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