run a "sip debug" and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW
Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru <cristi_icon...@yahoo.com> wrote: > Hello all, > > I have some issues with the MeetMe application. > > The working topology is as follows. The Asterisk (1.4.22-rc5) is connected > through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco > Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are > forwarded to Asterisk by the CM. > > The problem is that some users who are calling in from PSTN are getting > disconnected from the conference room after a period of time. They can get > in but after a while suddenly they are disconnected. The funny thing is that > on the Asterisk CLI/logs no errors/retrans/etc. appeared. > > The Asterisk has no Zaptel hardware. All the necesary modules are installed. > > Thanks, > Christian > > _______________________________________________ > -- 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