[asterisk-users] Conference problem

[Martin]

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
>
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