Steve Foy wrote:
Right... It just happened there now, this came up:
Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response)
So did it drop a few seconds into the call...like 5 - 15 seconds? If so
then you are having a problem with call setup. I would guess it is the
ACK that is not receiving a STATUS 200 OK so Asterisk cuts off the call.
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
[EMAIL PROTECTED]
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 <[EMAIL PROTECTED]>:
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
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Steve Foy | http://www.unite.net
UNITE Solutions | Tel: 028 9077 7338
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