asterisk users - Oct 2006 - Asterisk hangs up on incoming analog calls after a while

I have been experiencing a problem where after someone calls me from  
an analog line, the phone call is terminated after a period of time  
(anywhere from 15 seconds to 15 minutes)  The phone that I use to  
answer the call is an Aastra 9133i SIP phone.  There are several  
other SIP extensions on the network as well as a few analog  
extensions on a shared FXS line.  When a call comes in the analog  
line on the FXO, * dials all the extensions (SIP and analog.)  I have  
a Digium card with 1 FXO and 1 FXS.

How can I diagnose this problem?  Has anyone experienced anything  
similar?

Robert La Ferla wrote:
> I have been experiencing a problem where after someone calls me from an 
> analog line, the phone call is terminated after a period of time 
> (anywhere from 15 seconds to 15 minutes)  The phone that I use to answer 
> the call is an Aastra 9133i SIP phone.  There are several other SIP 
> extensions on the network as well as a few analog extensions on a shared 
> FXS line.  When a call comes in the analog line on the FXO, * dials all 
> the extensions (SIP and analog.)  I have a Digium card with 1 FXO and 1 
> FXS.

Do you have callprogress=yes or busydetect=yes in your 
/etc/asterisk/zapata.conf ?

On Oct 19, 2006, at 3:00 PM, asterisk-users-request@lists.digium.com  
wrote:
> Date: Thu, 19 Oct 2006 09:30:38 -0500
> From: "Eric \"ManxPower\" Wieling"
<eric@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@lists.digium.com>
> Message-ID: <45378C0E.7070708@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

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On Fri, 2006-10-20 at 22:38 -0700, Robert La Ferla
<robertlaferla@comcast.net> wrote:
> On Oct 19, 2006, at 3:00 PM, asterisk-users-request@lists.digium.com
> wrote: 
> > Date: Thu, 19 Oct 2006 09:30:38 -0500
> > 
> > From: "Eric \"ManxPower\" Wieling"
<eric@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@lists.digium.com>
> > 
> > Message-ID: <45378C0E.7070708@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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