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
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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_iconaru at 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 >
The CM is sending the BYE messages. ? Any ideas? ? Christian --- On Wed, 4/22/09, Martin <asterisklist at callthem.info> wrote: From: Martin <asterisklist at callthem.info> Subject: Re: [asterisk-users] Conference problem To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Date: Wednesday, April 22, 2009, 8:08 PM 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_iconaru at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/95633aea/attachment.htm