[Asterisk-Users] Sip call hang up

[Eric Wieling]

set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf

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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-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

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