[Asterisk-Users] Calls dropping off

[Rich Adamson]

Steve,
Since I have a rather short memory and receive about 250 posting per day, I
don't have a clue what has/hasn't been suggested. Here's a couple:
1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and
   and hints relative to the dropped calls
2. look at /var/log/asterisk/messages for hints
3. if the problem occurs frequently enough, start a ping from the * box to
   one or more of the sip phones to verify you're not loosing net connections
   at the time of the dropped call (Spanning Tree Protocol can mess with your
   infrastructure without you knowing it, as one example)
4. look in /var/log/asterisk/cdr-csv/Master.csv file to see if any hints in
   the cdr data
5. post a relavent definition from sip.conf so we have a clue how you've 
   defined a phone, as well as a relative Dial section from extensions.conf
   and zapata.conf 
6. I don't recall which sip phones you're using, but some have internal
   logging capabilities. If your's do, turn it on and look for hints.
7. Download ethereal and sniff the asterisk nic interface, ensure you stop 
   it right after a failure. If you need help doing the protocol analysis,
   then let me know.

Rich

------------------------
> I would have thought that if that was the problem, we couldn't makle or
> receive calls at all, or that we at least couldnt use all 3 Zap cards at the
> same time, but we can.
> 
> The problem only happens every so often, but recently it's getting more and
> more frequent... management are starting to get pissed :/
> 
> No more ideas?
> 
> I've tried everything else people have mentioned.
> 
> Cheers,
> Steve
> 
> On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote:
> > Hi,
> > 
> > Have you checked for IRQ conflicts ?
> > 
> > -b
> > 
> > Quoting Steve Foy <steve at unite.net>:
> > 
> > > Hi,
> > > 
> > > On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote:
> > > > Steve, 
> > > > 
> > > > this really is a FAQ. You need add to EACH (!) sip user something like
> > > > 
> > > > disallow=all
> > > > allow=ulaw
> > > > allow=alaw
> > > > allow=gsm
> > > 
> > > I do have that in my sip.conf. I am using ulaw.
> > > 
> > > Calls from the SIP phones through Asterisk and out one of my X100P cards are
> > > working 95% of the time and also, incoming calls through the X100P cards to
> > > the SIP phones are the same.
> > > 
> > > The only problem is that every once in a while, without any odd circustances
> > > that I can see, the call just drops and the remote user is gone.
> > > 
> > > The box running asterisk isn't under heavy load, so I can't see why this is
> > > happening.
> > > 
> > > I am not using g.729 or 723, just plain old ulaw, which I have got enabled
> > > in
> > > sip.conf
> > > 
> > > Cheers,
> > > Steve
> > > 
> > > -- 
> > > Steve Foy        |  http://www.unite.net
> > > UNITE Solutions  |  Tel: 028 9077 7338 
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> -- 
> Steve Foy        |  http://www.unite.net
> UNITE Solutions  |  Tel: 028 9077 7338 
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