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 wh
s
Dan
Sent from my Windows Mobile® phone.
-Original Message-
From: Bob Smither
Sent: 02 May 2010 14:04
To: Asterisk Users Mailing List - Non-Commercial Discussion >
Subject: Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
How can
14:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
> How can i log a continuous ping test to a file and include the date
> and time of each ping?
Try this:
#!/bin/
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
> 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
> 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 lo
ed Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: d...@keshercommunications.com
To: asterisk-users@lists.digium.com
Date: Thu, 29 Apr 2010 16:33:06 -0400
Subject: [asterisk-users] Calls Dropping
Hi,
I’m having a major problem with random calls dropp
Hi,
I'm having a major problem with random calls dropping. After spending weeks
trying to figure it out, i've finally spotted the issue but don't know how to
resolve it.
I run a sip server that's hosted in a data centre. It has a public IP address
with no nat involved. My provider also has a p
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
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 re
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
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?
It happens randomly once or twice a day to different callers.
Many thanks
Dan
___
-
sts.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 dropping
On Thu, Jun 25, 2009 at 9:55 PM, John Regal wrote:
> When using this method, it appears that the call file cre
On Thu, Jun 25, 2009 at 9:55 PM, John Regal 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.
> What I am exp
Hi,
I am using a call file formated like this:
Channel: local/12125557...@outbound/n
Callerid: 12125551212
Context: detect
Extension: s
Priority: 1
This sends the call into the dialplan at the [outbound] context. In
[outbound], I have:
[outbound]
exten => _1.,1,Dial(SIP/${ext...@flowroute,43)
If th
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. Th
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 usually
happens between
Hello,
I have scoured google for the last couple of days, implemented some changes but my issue is still occuring.
My company uses a hardware Bridge System for conferencing. Typically,
users will call in from cell phones but three always call from the VoIP
system.
Once or twice a day, one of the
Hi Guys,
I have a really odd one here.
We are dropping calls occasionally... there are no error messages being
spat out, but I can see this suspicious behavior in the debug logs;
Jun
30 14:58:48 DEBUG[19856] pbx.c: Function result is 'Other'
Jun 30 14:58:48 DEBUG[19856] pbx.c: Function resul
Ok,
I have had problems with calls dropping repeatedly today,
does anyone have any suggestions on what to make sure is not running? I have
x-windows disabled and Apache disabled. I noticed that mpg123 always seems to
have 2 processes running, is there any way to drop this down to just 1? A
> 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
p from time 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
]
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 PROT
sday, February 10, 2004 9:23 AM
> 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
>
ry 09, 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 an
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
d
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 wit
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
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.
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 stu
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/
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 pi
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
Hi!
> It's also showing up on the wiki:
> http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Where? ;->>
Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upda
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 mo
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 mo
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
>
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
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
>
>
> ___
> Asterisk-Users mailing list
>
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 somethi
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 throug
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 se
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/lis
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 -0
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 addin
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
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
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
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 t
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, duri
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.
Ther
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.
There is nothing in
51 matches
Mail list logo