<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;"><DIV>The CM is sending the BYE messages.</DIV>
<DIV> </DIV>
<DIV>Any ideas?</DIV>
<DIV> </DIV>
<DIV>Christian<BR><BR>--- On <B>Wed, 4/22/09, Martin <I><asterisklist@callthem.info></I></B> wrote:<BR></DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid"><BR>From: Martin <asterisklist@callthem.info><BR>Subject: Re: [asterisk-users] Conference problem<BR>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><BR>Date: Wednesday, April 22, 2009, 8:08 PM<BR><BR>
<DIV class=plainMail>run a "sip debug" and check whether it's asterisk disconnecting the<BR>calls (usually a SIP BYE message)<BR>or whether Asterisk is getting the disconnect from your Cisco GW<BR><BR>Martin<BR><BR>On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru<BR><<A href="http://de.mc1103.mail.yahoo.com/mc/compose?to=cristi_iconaru@yahoo.com" ymailto="mailto:cristi_iconaru@yahoo.com">cristi_iconaru@yahoo.com</A>> wrote:<BR>> Hello all,<BR>><BR>> I have some issues with the MeetMe application.<BR>><BR>> The working topology is as follows. The Asterisk (1.4.22-rc5) is connected<BR>> through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco<BR>> Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are<BR>> forwarded to Asterisk by the CM.<BR>><BR>> The problem is that some users who are calling in from PSTN are getting<BR>> disconnected from the conference room after a period of
time. They can get<BR>> in but after a while suddenly they are disconnected. The funny thing is that<BR>> on the Asterisk CLI/logs no errors/retrans/etc. appeared.<BR>><BR>> The Asterisk has no Zaptel hardware. All the necesary modules are installed.<BR>><BR>> Thanks,<BR>> Christian<BR>><BR>> _______________________________________________<BR>> -- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR>><BR>> asterisk-users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>><BR><BR>_______________________________________________<BR>-- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR><BR>asterisk-users
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