On Fri, 2006-10-20 at 22:38 -0700, Robert La Ferla <[EMAIL PROTECTED]> wrote: > On Oct 19, 2006, at 3:00 PM, [EMAIL PROTECTED] > wrote: > > Date: Thu, 19 Oct 2006 09:30:38 -0500 > > > > From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> > > > > Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog > > > > calls after a while > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > <[email protected]> > > > > Message-ID: <[EMAIL PROTECTED]> > > > > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > > > > > > Do you have callprogress=yes or busydetect=yes in your > > > > /etc/asterisk/zapata.conf ? > > > > > > No. They are not set. i.e. default
Let me guess: the incoming caller gets connected to the calling party via a Dial(Zap/1&Sip/what,...) type thing, and the Zap line answers and gets the call? and the Sip/what phone isn't even there? You merrily talk away and bang! you get disconnected not very long into your conversation? You should publish your console/log messages in those moments before and at the time of the hangup. I'll bet that some Sip phone in the Dial list has some event right when you hang up. See if you can narrow down your alternate dialing list to just the Zap and the Sip that are involved. Sounds like you may have a few different Sip phones involved. When you get it, then file a bug report. murf
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