[Asterisk-Users] Calls dropping off
Steve Foy
steve at unite.net
Thu Feb 5 07:39:47 MST 2004
Right... It just happened there now, this came up:
Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call 73f2d12d3ab7dc271e41f3e51990ff11 at 212.3.160.87 for seqno 3 (Response)
I'm not sure if that's related to it, but it's the only thing that came up
when the call got cut off.
Here's the generic sip.conf stuff
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all
allow=GSM
allow=G729
allow=iLBC
allow=SpeeX ; Allow all codecs
allow=ulaw
Here's a sip.conf declaration:
; Andy
[108]
type=friend
username=********
secret=********
host=dynamic
dtmfmode=rfc2833
callerid="Andy McAlister" <108>
context=internal
mailbox=108 at default
qualify=yes
canreinvite=no
And the relevant extension.conf bit:
;Andy
exten => 108,1,Dial(SIP/108,15)
exten => 108,2,Playback(int-voicemail/108)
exten => 108,3,Voicemail(s108)
exten => 108,102,Playback(int-voicemail/108)
exten => 108,103,Voicemail(s108)
Any insight vastly appreciated!
Cheers,
Steve
On Thu, Feb 05, 2004 at 06:33:06AM -0600, Rich Adamson wrote:
> 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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> > 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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