The problem is that i have random hangup in calls in the PSTN. After that I check in asterisk -rvvvvvv Sip show channels
And I see the extension.... The only way that I can place another call in the extension was to restart the Asterisk. -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Moises Silva Enviado el: Miércoles, 16 de Enero de 2008 09:31 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Problem with a channel And the problem is? ... I think you should read this: http://catb.org/~esr/faqs/smart-questions.html Regards, Moisés Silva On Jan 16, 2008 6:42 PM, Ruben Zamora <[EMAIL PROTECTED]> wrote: > > > > > I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5 > > > > I having these problem : > > > > Zap/2-1 is busy > > Hangup ZAP/2-1 > > Everyone is busy/congested at this time (1:1/010) > > Autofallthrough channel "SIP/202-b7b08ab0" Status is busy. > > > > And then HANGUP. > > > > > _______________________________________________ > -- 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 > -- "Within C++, there is a much smaller and cleaner language struggling to get out." _______________________________________________ -- 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