[asterisk-users] Re: Asterisk hangs up on incoming analog calls after a while

[Steve Murphy]

On Fri, 2006-10-20 at 22:38 -0700, Robert La Ferla
<robertlaferla at comcast.net> wrote:
> On Oct 19, 2006, at 3:00 PM, asterisk-users-request at lists.digium.com
> wrote: 
> > Date: Thu, 19 Oct 2006 09:30:38 -0500
> > 
> > From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
> > 
> > Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog
> > 
> > calls after a while
> > 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > <asterisk-users at lists.digium.com>
> > 
> > Message-ID: <45378C0E.7070708 at fnords.org>
> > 
> > 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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