Re: [Asterisk-Users] Sip call hang up
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 <[EMAIL PROTECTED]> 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
On Wed, 15 Oct 2003 14:54:49 -0500 Eric Wieling <[EMAIL PROTECTED]> wrote: > 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. > Ok, I got it. But, could it be the cause of my problem, since the default is 'no'? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
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 <[EMAIL PROTECTED]> 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 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
On Wed, 15 Oct 2003 11:16:03 -0500 Eric Wieling <[EMAIL PROTECTED]> 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
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 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip call hang up
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
