[Asterisk-Users] Sip call hang up

[Eduardo Goncalves]

I did these modfications, but the problem persist. After some minutos
the sip calls hang-up. :~

Eduardo

> 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