On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote:
Hi Bob,
Thanks for that. Is there any way I can make the task run in the
background and free up the console? Also so that I can disconnect my
ssh session without losing the task.
Thanks
Dan
Matthieu NICAISE mentioned screen which
my advise check your internet connection on the remote location and keep a
ping from that network to your server running all the time to check for time
outs.
How can i log a continuous ping test to a file and include the date and time of
each ping?
I've found this bash code but it only logs
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
snip
How can i log a continuous ping test to a file and include the date
and time of each ping?
Try this:
#!/bin/sh
for (( ; ; ))
do
NOW=$(date +%T %m/%d/%Y)
PING=$(ping -qc 1 example.com)
echo $NOW: $PING pinger.log
done
exit 0
Sent: 02 May 2010 14:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
snip
How can i log a continuous ping test to a file and include the date
Sent from my Windows Mobile® phone.
-Original Message-
From: Bob Smither smit...@c-c-i.com
Sent: 02 May 2010 14:04
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34
i'm having the same problem with one of my call centers located in Egypt..
although the ip-phones are located on a Dedicated Leased Line yet calls drop
out of the blue.almost an identical setup as yours..provider located in France
(data center) my server located in Sweden (data center) both on
The info you need is here
http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf
Ish
Dan Journo wrote:
Hello,
We have a problem that calls seem to be dropping for no reason.
Is there any way to write a debug log to disk so that I can check it
as soon as a call is lost?
On 11 Dec 2009, at 11:19, Dan Journo wrote:
Is there any way to write a debug log to disk so that I can check it
as soon as a call is lost?
It happens randomly once or twice a day to different callers.
/var/log/asterisk/full?
Most 'standard' setups produce it. Failing that google will
Thanks.
I didnt stop that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: 11 December 2009 11:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
%3D1234512345
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, June 26, 2009 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls
On Thu, Jun 25, 2009 at 9:55 PM, John Regaljre...@gmail.com wrote:
When using this method, it appears that the call file creates the first part
of the call, then creates a second call with the Dial() app. Once the call
executed by the Dial() app is answered, the two calls are joined together.
Carlos Chavez wrote:
I have a customer that recently started having a problem with their
Call Center SIP extensions. The problem is that after some time the
caller will hear a triple tone (beep, beep, beep), a 5 second pause,
another triple tone and then the call will be dropped. This
Of Ejay Hire
Sent: Tuesday, February 10, 2004 5:46 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Calls dropping off
I have this problem intermittently, and doing an asterisk -r showed
too many retries. hunting around with ethereal found a bad hub.
-e
, 2004 3:35 PM
To: Michael Nigrelli
Cc: Asterisk-Users
Subject: Re: [Asterisk-Users] Calls dropping off
On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote:
Steve,
Did you ever figure out why this happens. I have had asterisk up and
running for a few weeks and all of a sudden
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Calls dropping off
Last 2 days I have noticed that more and more often calls
are
just being
dropped. I can't find any logs or anything indicating that
something is
wrong. If I do a trace and wait for a call to drop I can
only
see
]
Subject: RE: [Asterisk-Users] Calls dropping off
I have this problem intermittently, and doing an asterisk -r showed
too many retries. hunting around with ethereal found a bad hub.
-e
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
Tomica
to time.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire
Sent: Tuesday, February 10, 2004 5:46 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Calls dropping off
I have this problem intermittently, and doing an asterisk -r
On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote:
Steve,
Did you ever figure out why this happens. I have had asterisk up and
running for a few weeks and all of a sudden this started happening.
Exactly the same here, it was running fine for about a month or so. Then one
day,
On Fri, Feb 06, 2004 at 08:18:21PM -0500, Andres wrote:
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
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
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
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)
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
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)
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.
Still no luck, calls are still dropping off about the same amount as before.
Any more ideas!?
On Fri, Jan 30, 2004 at 05:14:27PM +, Steve Foy wrote:
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote:
Try adding it to
Steve Foy wrote:
Still no luck, calls are still dropping off about the same amount as
before.
Any more ideas!?
On Fri, Jan 30, 2004 at 05:14:27PM +, Steve Foy wrote:
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel
Hi!
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Ta
SJ
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On Mon, Feb 02, 2004 at 05:28:38PM +0100, Philipp von Klitzing wrote:
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Didn't think so.
I don't understand why this is happening, the server isn't
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
Do you have busydetect=yes and/or callprogress= in zapata.conf? If so
set them to no.
On Mon, 2004-02-02 at 11:10, Steve Foy wrote:
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
Philipp von Klitzing wrote:
Hi!
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Ta
SJ
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[EMAIL PROTECTED]
On Mon, Feb 02, 2004 at 11:16:16AM -0600, Eric Wieling wrote:
Do you have busydetect=yes and/or callprogress= in zapata.conf? If so
set them to no.
I did have busydetect=yes, I've just commented it out.
I don't see that busydetect would cause problems, as the call does get
answered and could
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
On Mon, 2004-02-02 at 11:21, Senad Jordanovic wrote:
Philipp von Klitzing wrote:
Hi!
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Well...
Googling... Few months ago produced that
On Mon, 2004-02-02 at 11:21, Senad Jordanovic wrote:
Philipp von Klitzing wrote:
Hi!
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Well...
Googling... Few months ago produced that
Hi!
It's also showing up on the wiki:
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Where? ;-
Philipp
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Philipp von Klitzing wrote:
Hi!
It's also showing up on the wiki:
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Where? ;-
Philipp
Interesting...!
Mysteriously... reinvite has EDITED it self in above URL to
canreinvite in space in few hours... :)
Ta
SJ
Steve Foy wrote:
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.
Shot in the dark here ...
Do you have:
canreinvite=no
Set in sip.conf for the SIP phones in question ?
Ciao,
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a
On Fri, Jan 30, 2004 at 01:18:29PM +0100, Olle E. Johansson wrote:
Enable 'sip debug' at the CLI and send some detailed log file.
It's very difficult to catch the logs when this happens, it doesn't happen
all the time, and I'm hardly ever on the phone so, it would be even less
likely to happen
Bill,
On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
Shot in the dark here ...
Do you have:
canreinvite=no
Set in sip.conf for the SIP phones in question ?
No, I don't.
All I have in sip.conf is the general stuff like:
[general]
port = 5060 ; Port to
Try adding it to the phones involved so it looks like this:
; Shirley
[100]
type=friend
username=xxx
secret=xxx
host=dynamic
dtmfmode=rfc2833
callerid=Shirley O'Neill 100
context=internal
[EMAIL PROTECTED]
qualify=yes
canreinvite=no
-b
Quoting
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote:
Try adding it to the phones involved so it looks like this:
; Shirley
[100]
type=friend
username=xxx
secret=xxx
host=dynamic
dtmfmode=rfc2833
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