2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com
Greetings.
I got two extensions on my asterisk that autenticates from outside our
network, via internet. Is there a way to monitor, in certain time periods,
if they are available (online) and send some sort of notification if they
Top and bottom post in the same email... don't open again the thread :-)
#!/bin/bash
res=`sudo /usr/sbin/asterisk -rx 'sip show peer $1' | grep Status | cut
-d\: -f 2 | cut -d\ -f 2`
if [ $res == OK ]
then
echo OK is registered
exit 0
else
echo WARNING peer not registered
exit 1
2013/1/8 Luis
2012/10/22 Binan AL Halabi binanalhal...@yahoo.com
Hi,
You are using b flag in monitor command. This means don't begin recording
untill call is bridged.
So what you get if you delete this flag ?
If I dont use the b flag then I get two separate files just like in the
case when B waits till
Grzegorz Pycia wrote:
Hi
I have some problem with monitor application when call i transferred
in
attended mode and the transfer occurs before call is answered.
Here is how it looks:
A calls B(let's assume ${UNIQUEUEID}=1)
exten = _,1,NoOp
seme =
I'm using latest 1.8, althought I did check and this behaviour is the same
since 1.6.2.11. I will file a bug report about it in 1.8.17.0.
Auto Mixing would not bother me, i will check the Mix monitor.
Regards.
22 paź 2012 17:22, Jonathan Rose jr...@digium.com napisał(a):
Grzegorz Pycia wrote:
Hi,
You are using b flag in monitor command. This means don't begin recording
untill call is bridged.
So what you get if you delete this flag ?
// Binan
Från: Grzegorz Pycia grzegorz.py...@thulium.pl
Till: asterisk-users@lists.digium.com
Skickat: lördag,
Suggestion 1 - mixmonitor instead of monitor
Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
: [asterisk-users] Monitor Command Records separate channales
Suggestion 1 - mixmonitor instead of monitor
Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday
] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter
Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter
: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
problem is not mixing It's the same file but inside that file two
seperate records first callers then reciever
Sent from my
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠
List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny
Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two
] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue
Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I would wager that your setup dumps what would normally be in /v/l
Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
[root@c-24-1-71-68 asterisk]# grep -R 'monitor' *
From
Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk
-users awaits moderator approval
Your mail to 'asterisk-users' with the subject
RE: [asterisk-users] Monitor Command Records separate channales
Is being held until the list moderator can review it for approval.
The reason it is being held:
Message body is too big: 1004233 bytes with a limit
Un-top-posting, snarky comments inline...
On Wed, 28 Dec 2011, Faraj Khasib wrote:
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation and
receiver its single file but separate recording, How can I make
Hello,
Do you use monitor?, because in asterisk 1.4 to new versions, It's use
mixmonitor, in asterisk 1.2 had this mistake.
Regards
On Wed, Dec 28, 2011 at 10:11 PM, Steve Edwards
asterisk@sedwards.comwrote:
Un-top-posting, snarky comments inline...
On Wed, 28 Dec 2011, Faraj Khasib
Once the call is completed you can use SOX to split the call. In my
opinion, you will have to get a larger ram disk or record the files to a
different format like WAV49, but maybe somebody has a better solution for
you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi.
Yeah, sox and soxmix are no problem - we're already using that to
merge/join all of the segments together if people pause then resume the
recordings mid call.
The main issue is getting Asterisk to split the recordings into segments
even when users don't pause/resume the recordings (which
Hi,
I've done some similar thing in one of my testing, using MixMonitor and
monitor at the same time. Everything worked perfectly well no issues even on
Vmware. Can you check if the CPU utilization is normal. Also which version
of asterisk you are using?
--
Regards,
Sammy
On Thu, Oct 20, 2011
Hi,
CPU usage does not change when a call is served by Asterisk. I performed
several there was no influence.
Version is Asterisk 1.6.2
Regards,
Date: Thu, 20 Oct 2011 13:30:20 +0500
From: Sammy Govind govoi...@gmail.com
Subject: Re: [asterisk-users] Monitor does not work well (little cuts
2011 13:30:20 +0500
From: Sammy Govind govoi...@gmail.com
Subject: Re: [asterisk-users] Monitor does not work well (little cuts
in the audio file)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
cajujwtg
Asterisk-SNMP could be an option for u.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Friday, June 24, 2011 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Monitor
man, see monast http://monast.sourceforge.net/
--
Renato dos Santos
shazaum.wordpress.com
2010/8/17 Matt Riddell li...@venturevoip.com
On 17/08/10 6:34 PM, Hans Witvliet wrote:
On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote:
Might be worth your time to check out:
Look into astassisstant. I don't remember the website, but google will take
you their. It doesn't need any installations on the server, just a manager
user in manager.conf.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-18 8:39 AM, Shazaum shaz...@gmail.com wrote:
man, see monast
On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote:
Might be worth your time to check out: http://www.humbuglabs.org/
Though they write:
...
insight into the enterprise’s telephony infrastructure. Utilizing a set
of none-intrusive analytical technologies, Humbug is capable of
On 17/08/10 6:34 PM, Hans Witvliet wrote:
On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote:
Might be worth your time to check out: http://www.humbuglabs.org/
Though they write:
...
insight into the enterprise’s telephony infrastructure. Utilizing a set
of none-intrusive
Might be worth your time to check out: http://www.humbuglabs.org/
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Saturday, August 07, 2010 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
maybe, can use the sneplivre for this...
www.sneplivre.com.br
detail is that in portuguese, this can be translated easily (i think)
Renato dos Santos
ren...@opens.com.br
OpenS Tecnologia Ltda
Rua Padre Marcelino Champagnat, 236
Jardim Atlântico - Florianópolis - SC - Brasil
+55 (48) 3954-8000
Hallo Keane,
I truly have a nagios server, up and running 24/7
--
Richard Zulu
Managing Director
Time Information Company
P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug
Mobile :+256752624006
Skype: zulu.richard
--
Thanks Nasri,
I don't want to only be able to use the CLI because I need the Helpdesk and
application support Unit to be able to monitor, and they are not all the
techy with CLI and stuff..
On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal na...@ictinnovations.comwrote:
Hi
following asterisk
-Commercial Discussion
Subject: Re: [asterisk-users] Monitor asterisk
Thanks Nasri,
I don't want to only be able to use the CLI because I need the Helpdesk and
application support Unit to be able to monitor, and they are not all the techy
with CLI and stuff..
On Sun, Aug 8, 2010 at 5:00 AM, Nasir
I agree with you and suggest you to use CLI command via AMI, for example
Command core show channels
I prefer CLI commands when they are available, as they return an aggregate
response as compared to AMI you do not need to filter, identity, and group
multiple responses / events to get result of a
Hi
following asterisk cli commands can help
show channels, show uptime and show sysinfo
here is an example
asterisk -x core show sysinfo
On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugwrote:
Hey guys,
I have my asterisk box running without a gui. I now need to
yes, sox is installed.
Anyway, I changed the lines that read: Monitor(gsm,/var/log) to
MixMonitor(/var/log/file.gsm...)
Thanks for answering.
2009/12/16 Holger von Ameln holger.von.am...@peercom.net
This may be pretty obvious but do you have sox installed? I managed to
forget that on
This may be pretty obvious but do you have sox installed? I managed to forget
that on more than one occasion ;-)
--
Holger von Ameln
Peercom Ltd. Co. KG
holger.von.am...@peercom.net
Tel.: +49 (0) 511-84887106
http://www.peercom.net/peercomshop
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dáibhéad Antoine O'Reilligh wrote:
Have I forgotten anything?
Do you have 'sox' installed on your asterisk box?
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
I just solved this :)
Turns out that was removed in Asterisk 1.6 :) The solution is:
monitor-type = MixMonitor
monitor-format = wav
Hopefully will help somebody!! also has a really neat after-recording shell
command option, so after a file has been saved you can instantly have it
converted to
if you haven't exectued the queue cmd you cannot know who will took that call.
You cannot know this before the agent took it because there are many agents who
can do it.
You can know it via cdr or manager interface, but only when the call is tooked
or finished.
On Mon, Jul 06, 2009 at 03:23:29PM
On 21 May 2009, at 22:02, Nikhil Nair wrote:
I'm pretty stumped here; I can only imagine that, for some reason,
not all
silence is being recorded in the sound files.
Silence suppression might be enabled somewhere? Asterisk doesn't like
that generally, so might screw recordings too..
Steve
(monitor legs are out of sync)
On Thu, 21 May 2009, Nikhil Nair wrote:
I'm running Asterisk 1.2.13...
A more modern version wouldn't hurt.
I've been using monitor() to record calls, with fairly satisfactory
results - at least until the last few months.
If you don't need the legs separate,
Hi Nikhil,
Several of these out of sync issues have been resolves in many recent
versions
of Asterisk. I'm not sure if many of the out of sync issues were reported
against 1.2 when it was receiving bug updates, so you may need to move to
Asterisk 1.4 in order to get these updates.
The problem in this particular case is that the actual monitor object is on
A's
channel. When A is no longer involved in the call, the monitor is gone, and
so
the call cannot be recorded further. One possible solution is to run the
Monitor
application on B's channel instead. This
Gunnar Schaller wrote:
Hello list,
I need to record all calls. So I'm using application Monitor. Works
good until someone transfers a callee to another internal extension.
Example:
A calls B
A set B on hold
A calls C
A transfers B to C with SIP transfer (SIP REFER - with phone funktions
Hi Paolo,
You can always supply a command with MixMonitor to rename the file after
the call is completed. At that point the variables are evaluated and you
can probably rename/move the recording to which phone answered the call.
It looks like Monitor doesn't have this feature, but I'm sure you
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote:
One of the box that have Asterisk 1.4.18 is properly merging calls and
the other box that has Asterisk 1.4.15 is recording the calls but not
merging them, I have made sure that SOX is installed on the box.
It might be worth giving the
Newer version of sox don't seem to have soxmix anymore, but you can
use sox -m and I think asterisk should be changed to use that instead.
on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote:
One of the box that have Asterisk
PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai,
New Delhi
Subject: Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore
, New Delhi
Subject: Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore, but you can
use sox -m and I think asterisk should be changed to use that instead.
on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote
On Mon, 2008-04-21 at 21:11
], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai,
New Delhi
Subject: Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore
MixMonitor.
And please stop posting the same question to the list over and over.
Sanjay Rajdev wrote:
What is good for recording all the incoming and outgoing calls,
Monitor() or MixMonitor().
Regards,
Sanjay Rajdev
You can try using the asterisk -r -x CLI command
This allows you connect to the asterisk on the machine u run the command.
As for APIs have I have no ideas. May be the seniors can help you.
Thank you.
___
-- Bandwidth and Colocation Provided by
14 feb 2008 kl. 22.35 skrev Benny Amorsen:
Matthew J. Roth [EMAIL PROTECTED] writes:
Yes, asterisk -rx will only allow you to execute CLI commands. It
also tends to spew out a bunch of garbage that makes parsing
difficult
unless verbosity is always set to 0.
It would be very handy if
On Fri, Feb 15, 2008 at 08:55:11AM +0100, Johansson Olle E wrote:
I would also like to see manager wrappers that produce data that is
easy to handle for scripts, like a wrapper that produces show channels
consise in various formats. Do we have a perl programmer on
the list?
Such a generic
Johansson Olle E wrote:
In the long run we're trying to move to using the manager for all
parsing by adding a lot of new manager events and actions.
If there's something missing that you only can do or information you
only can get in the CLI, please tell us.
Olle,
How does what you are
Soumya Kat wrote:
Thank you to all those who replied to my last query. For them and for
the suggestion, I can monitor asterisk using the asterisk -r -x
command option. What I would like to know is that using asterisk
-r -x way I can only use the *CLI commands. Is there any other way in
Matthew J. Roth [EMAIL PROTECTED] writes:
Yes, asterisk -rx will only allow you to execute CLI commands. It
also tends to spew out a bunch of garbage that makes parsing difficult
unless verbosity is always set to 0.
It would be very handy if it was possible to turn off messages that
aren't
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Soumya Kat wrote:
Thank you for replying. The probleam is how do I use the
Asterisk_manager-API and implement them in my C code. Like how do I call a
API in my C program. It will be of great help if I can have an example.
By traffic I mean how
Thank you for replying. The probleam is how do I use the
Asterisk_manager-API and implement them in my C code. Like how do I call a
API in my C program. It will be of great help if I can have an example.
By traffic I mean how much bandwidth or data transferring is taking place in
a call that is
Soumya Kat wrote:
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and
Soumya Kat wrote:
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and
On Wed, 2007-08-29 at 10:46 -0400, Nitesh Divecha wrote:
Anyone using AGI scripts to monitor their systems?
Something like if the system goes down, AGI script will be triggered and
system admin will be notified saying System XYZ has gone down...
If the system goes down, how would an AGI
Thanks Jared,
Basically, it would be a totally different system running Asterisk with
AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
specifically monitoring ports (80, 21, 25) but whole system. If system
timeouts then AGI scripts are triggered and notify system admin.
On 8/29/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Basically, it would be a totally different system running Asterisk with
AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
specifically monitoring ports (80, 21, 25) but whole system. If system
timeouts then AGI scripts
Thanks man,
That's what I was looking right now... to use Nagios with asterisk plug-ins.
Cheers,
Nitesh
James FitzGibbon wrote:
On 8/29/07, *Nitesh Divecha* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Basically, it would be a totally different system running Asterisk
with
Quoting Nitesh Divecha [EMAIL PROTECTED]:
Thanks Jared,
Basically, it would be a totally different system running Asterisk with
AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
specifically monitoring ports (80, 21, 25) but whole system. If system
timeouts then AGI
Diego Iastrubni wrote:
I am seeing on my logs this message:
Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160
on channel 3
Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160
on channel 3
(repeated much more then what I will show
On Wed, Aug 08, 2007 at 12:56:33PM +0300, Diego Iastrubni wrote:
Hi all,
I am seeing on my logs this message:
Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160
on channel 3
Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160
on
On Thursday 09 August 2007 09:40, Philipp Kempgen wrote:
Because you are logging debug messages :-P
---cut---
; Debug mode turns on a LOT of extra messages,
; most of which you are unlikely to understand without an understanding of
; the underlying code. Do NOT report debug messages as code
Diego Iastrubni wrote:
On Thursday 09 August 2007 09:40, Philipp Kempgen wrote:
So if there is no problem with your system just don't enable
debug mode.
See Tzafrir's reply, that would explain much more.
That message did not yet make it to me.
The list still seems to have delivery problems.
Stefan Reuter wrote:
Hey Daniel,
I think adding the events would be a good idea.
Just open an issue on http://bugs.digium.com/ and attach your patch
there. Be sure to send a disclaimer to digium so your patch can be
included in the distribution (see
Anthony Francis wrote:
Stefan Reuter wrote:
Hey Daniel,
I think adding the events would be a good idea.
Just open an issue on http://bugs.digium.com/ and attach your patch
there. Be sure to send a disclaimer to digium so your patch can be
included in the distribution (see
Dear Sir,
On Wed, 11 Jul 2007, Anthony Francis wrote:
WTF
I am intrigued by your ideas and would like to subscribe to your
quarterly newsletter.
-- Alex
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
Anthony Francis wrote:
I guess I should clarify. My name is Anthony, I was the one that said I
had written a patch, if Daniel also said he had done so and I missed
that email I apologize.
Well then disclaim it and post it to the asterisk bug tracker or post
its issue id if you already did.
Hey Daniel,
I think adding the events would be a good idea.
Just open an issue on http://bugs.digium.com/ and attach your patch
there. Be sure to send a disclaimer to digium so your patch can be
included in the distribution (see
http://asterisk.org/developers/bug-guidelines for details).
Daniel Gradecak wrote:
Hi all,
I would like to know if there is any possibility to send an event when a
call is monitored?
For both start and stop monitor.
There is no event sent on asterisk 1.2 for that monitor case. I did not
find any changes regrding that on 1.4. Am I wrong?
Is it
Hi Anthony,
are you sure the monitor is started and sotoped via the dialplan ?
Anthony Francis wrote:
Daniel Gradecak wrote:
Hi all,
I would like to know if there is any possibility to send an event when a
call is monitored?
For both start and stop monitor.
There is no event sent on
Anthony Francis wrote:
There are no events generated when the monitor stops and starts, but
since you are implicitly recording in your dialplan one way or another
you can just add a userevent step before recording and after.
You can also start monitoring through the Manager API in which case
On 7/9/07, Daniel Gradecak [EMAIL PROTECTED] wrote:
are you sure the monitor is started and sotoped via the dialplan ?
If you're using Monitor() or MixMonitor(), then just add a UserEvent() call
just before it in the dialplan.
If you're doing monitoring of queues, it's a bit trickier - you
Hi Stefan,
actually you probably know i am using your java-asterisk :)
Yes the best solution i found till now it was to add those events to
res_monitor.c. I wonder why it was not yet done, may be there was a reason
or nobody needed it yet.
Anyhow this would be a cool feature that others should
James FitzGibbon wrote:
On 7/9/07, *Daniel Gradecak* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
are you sure the monitor is started and sotoped via the dialplan ?
If you're using Monitor() or MixMonitor(), then just add a UserEvent()
call just before it in the dialplan.
Thanks for the answer Matthew.
I'm having high load, choppy sound and slow responsives with an
asterisk server (version 1.2.12.1) that make a peak of 90 channels
(around 60 phones calling at max, isn't necessary to reach this peak
to get the problem). All the traffic is SIP, with
On Tue, 29 May 2007, Edgar A. Luna Diaz wrote:
The real problem was found. The configuration of this server had a
recording path as /var/spool/asterisk/monitor/ for every call, so the
size of monitor (the directory) keeps growing at 2000 files per day. Its
peek was around 36MB, just containing
Edgar A. Luna Diaz wrote:
I'm having high load, choppy sound and slow responsives with an
asterisk server (version 1.2.12.1) that make a peak of 90 channels
(around 60 phones calling at max, isn't necessary to reach this peak
to get the problem). All the traffic is SIP, with recording for
So there are 0 watchers while the GXP is configured to that hint? are you
sure you set the phone to Asterisk BLF?
On 11/15/06, Ken Williams [EMAIL PROTECTED] wrote:
Upon further investigation I must be doing something wrong.
It was my understanding that a hint extension could be anything, it
Upon further investigation I must be doing something wrong.
It was my understanding that a hint extension could be anything, it
wasn't the same as a real extension, though you could make it the same
to make it easier.
That being said exten = 702,hint,SIP/702 works, while exten =
On Mon, 2006-10-16 at 08:00 -0500, Tim Connolly wrote:
Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686
running Linux on 2006-06-17
When I used monitor, I seem to get most calls cut off if they run
very long. Sometimes two minutes, sometimes 5 or 15.. Seems
Then you have something wrong. Inuse should always be 0 if the card is not in use. Each time a call is in process, the agi will change that +1 and when the agi is complete it will -1.
I don't think that FOP will work without a ton of modifications unless someone at A2Billing has a patch or
That doesn't seem to work, most of the cards have the inuse value set
to something different than zero, wheather they are in use or
not...anyway
Thanks
I'll keep digging
George
On 10/9/06, William Piper [EMAIL PROTECTED] wrote:
A2B already shows this in the DB. If you have any php/perl skills
On 10/10/06, George Masgras [EMAIL PROTECTED] wrote:
Hello all!I'm currently using Asterisk in conjunction with a2billing andeverything seems to be working great so far. Now, all I'm missing issome sort of a GUI to monitor all calls going out through my trunks. I
can always do 'sip show channels'
A2B already shows this in the DB. If you have any php/perl skills just run a query like the following:
select * from cc_card where inuse 0
Then write a js to refresh the script everyfew seconds. I believe this will give you the reporting that you'd need.
bp
On 10/9/06, George Masgras [EMAIL
Bad form to reply to your own post, but even worse form when you can't
read the screen.
Try StopMixMonitor :)
Julian
Julian Lyndon-Smith wrote:
Is there an equivalent stopmonitor command if you are using MixMonitor ?
StopMonitor does not seem to have an effect on MixMonitor
Julian.
Hi,
You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy.
Idris
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006
12:23 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitor
a particular SIP call
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