Hi, We are running Asterisk SVN. We are facing a strange and repetable problem. All outgoing call gets terminated in approx 20 minutes. Asterisk initiates BYE message to the remote end and call terminates. Can anyone help? Thanks Jim
Marcin J. Kowalczyk
2008-Oct-31 11:47 UTC
[asterisk-users] Call terminates after 20 minutes
Jim Boykin pisze:> We are running Asterisk SVN. We are facing a strange and repetable > problem. All outgoing call gets terminated in approx 20 minutes. > Asterisk initiates BYE message to the remote end and call terminates. >Sesion-timer set but not supported by sip-peers?
Marcin, can you elaborate. No timer has been set and call is not idle either. Thanks Jim On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk <marcin.kowalczyk at ccig.pl> wrote:> Jim Boykin pisze: >> We are running Asterisk SVN. We are facing a strange and repetable >> problem. All outgoing call gets terminated in approx 20 minutes. >> Asterisk initiates BYE message to the remote end and call terminates. >> > Sesion-timer set but not supported by sip-peers? > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Any help. Thanks On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin <boykinjim at gmail.com> wrote:> Marcin, can you elaborate. No timer has been set and call is not idle either. > > Thanks > Jim > > On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk > <marcin.kowalczyk at ccig.pl> wrote: >> Jim Boykin pisze: >>> We are running Asterisk SVN. We are facing a strange and repetable >>> problem. All outgoing call gets terminated in approx 20 minutes. >>> Asterisk initiates BYE message to the remote end and call terminates. >>> >> Sesion-timer set but not supported by sip-peers? >> >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >
Copy your dialplan and sip debug a call. On Sun, Nov 2, 2008 at 3:07 PM, Jim Boykin <boykinjim at gmail.com> wrote:> Any help. Thanks > > > On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin <boykinjim at gmail.com> wrote: > > Marcin, can you elaborate. No timer has been set and call is not idle > either. > > > > Thanks > > Jim > > > > On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk > > <marcin.kowalczyk at ccig.pl> wrote: > >> Jim Boykin pisze: > >>> We are running Asterisk SVN. We are facing a strange and repetable > >>> problem. All outgoing call gets terminated in approx 20 minutes. > >>> Asterisk initiates BYE message to the remote end and call terminates. > >>> > >> Sesion-timer set but not supported by sip-peers? > >> > >> > >> > >> _______________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Juan E. Rodr?guez Cel. 829-886-5565 Work: 809-724-9227 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081102/00ff7645/attachment.htm