Hi,
can you recommend way to test status of PJSIP endpoint (SIP trunk to the
operator)?
is there something better than parsing
asterisk -rx "pjsip show contact operator/sip:operator@1.1.1.1:5060"
?
We are using icinga2/prometheus
Marek
--
Interesting approach.
What we've done is to write an app that runs on a separate machine that
simply does some asterisk -rx calls to the running Asterisk instance via an
SSH library and then evaluate the string returned.
For example, to monitor our registered SIP service providers, we compare
Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
owners, How can we monitor asterisk, what happens if service stop
responding.
Here is a small howto on monitoring asterisk with nagios. I am sure there
are plenty of options and suggestions, but this is one of them and has
Hi
I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my
primary goes down. I'm wondering what the best method of checking if the
primary being up is.
Is DIALSTATUS suitable for this or is there any good SIP headers to look
at after the Dial step?
Thanks in Advance
Ish
--
Ishfaq
i think DIALSTATUS is not suitable for failover if trunk is down you get
dialstatus after time out in dial string.
it is too late for failover, you can use some script to check if
destination host is up.
if you want to do failover when destination host is up then dialstatus are
good.
On Tue, May
On 14/5/13 4:30 pm, Ishfaq Malik wrote:
I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my
primary goes down. I'm wondering what the best method of checking if the
primary being up is.
Well, the obvious start point might be ChanIsAvail() - that'll at least
weed out an upstream
We use Zabbix as monitoring tool and SNMP to get statistics and other
info from Asterisk.
for this you will have to make sure the snmp module for asterisk gets
compiled and the Asterisk MIB is used.
Regards,
Michel.
On 09-05-13 21:23, motty cruz wrote:
Hello,
i'm looking for suggestions to
Try with http://www.observium.org (Observium).
You can customize script to report into Observium's dashboard.
Regards,
On Fri, May 10, 2013 at 7:55 AM, Michel Verbraak mic...@verbraak.orgwrote:
We use Zabbix as monitoring tool and SNMP to get statistics and other
info from Asterisk.
for
Hello,
i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
but no success, I do prefer not to install any web server on the server
running Asterisk.
Thanks in advance.
-Motty
--
_
-- Bandwidth and
Monitor what parts exactly?
Right this moment I'm in the process of installing Munin and the Asterisk
plugin to monitor channel usage, SIP connections, and the like. The Munin
server is running on a separate machine with just the node software on
Asterisk.
On Thu, May 9, 2013 at 12:23 PM,
Thanks for the suggestion Carlos,
do you have a HowTo? can you point me to one.
I unsuccessfully follow one found using google. I'm using CentOs 6.0
Thanks,
Motty
On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote:
Monitor what parts exactly?
Right this moment I'm
I'm using opennms and It's working fine.
On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:
Hello,
i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
but no success, I do prefer not to install any web server on the server
running Asterisk.
http://opennms.org/wiki/Installation:Yum
On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote:
I'm using opennms and It's working fine.
On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:
Hello,
i'm looking for suggestions to monitor Asterisk
It's not quick or simple, but there's decent documentation. I haven't been
saving the links I used, so I can't just give you specific places to look,
other than the best Asterisk plugin:
https://github.com/munin-monitoring/contrib/blob/master/plugins/asterisk/asterisk
TIP: Use chmod 755 on the
Thanks for your help; I just want to monitor the queue, calls on hold
average time, incoming out going call, I only want to monitor Asterisk, not
the server Asterisk in running on.
thanks,
-Motty
On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Then you want a queue manager and reporting tool. Usually when people say
monitor Asterisk is has to do with the state of the system itself. You
should look at http://www.asternic.net and similar products. Munin will
tell you channels in use, but not the other stuff you want.
On Thu, May 9,
You can use queue-stats
http://www.asternic.org/stats/demo/
they has a free version
On Thu, May 9, 2013 at 4:12 PM, motty cruz motty.c...@gmail.com wrote:
Thanks for your help; I just want to monitor the queue, calls on hold
average time, incoming out going call, I only want to monitor
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] monitoring Asterisk 1.8
Thanks for your help; I just want to monitor the queue, calls on hold average
time, incoming out going call, I only want to monitor Asterisk, not the server
Asterisk in running on.
thanks,
-Motty
On Thu
There is nagios plugin
check_asterisk_channels
Examples:
Check channels/calls, with no concern about limits.
check_asterisk_channels
Check channels/calls. Issue a warning if there are more than 10 active
channels, and a critical if there are more than 15 active channels.
How can I monitor channel that hangup?
I'm using asterisk 1.8.15.1 and there are many times that nobody is using the
line but when I run:
asterisk -rx core show channels it show:
Channel Location State Application(Data)
SIP/pstn--00 (None)
How can I monitor asterisk if all lines are registered etc?
I have an asterisk on a remote location and sometime they reporting problems
that phone is not ringing, they can not dial out etc.
Usually I just restart asterisk and it solves the problem.
Is there an application that will email me
=-
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
[syscon...@gmail.com]
Sent: Thursday, November 22, 2012 4:18 PM
To: Asterisk Users List
Subject: [asterisk-users] monitoring asteriks
How can I monitor
How about any of these programs listed in:
http://www.voip-info.org/wiki/view/Asterisk%2Bmonitoring
--
Joseph
On 11/22/12 17:04, Michelle Dupuis wrote:
take a look at AsteriskControl script at www.generationd.com
This is a free script that monitors, responds to IP address changes, etc. and
Hi,
I had a zabbix http://zabbix.org/wiki/Main_Page monitoring server with
zabbix_agent installed on the asterisk server. Zabbix server requests the
agent to execute an AMI script and pull information about the phone in the
given argument to that function. That AMI script returned the 1 from the
2012-04-09 22:32, Johan Wilfer skrev:
2012-04-09 20:22, Carlos Alvarez skrev:
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
At first, if your Asterisk is in a VM install it on the real
server, it solved us on some
On Wed, Apr 11, 2012 at 4:29 AM, Johan Wilfer li...@jttech.se wrote:
Sounds very reasonable. Do you run this on a dedicated server, and
configured the switch to duplicate the traffic to the quality server? Or do
you run this on the same server as asterisk?
Cheap dedicated server with a span
After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed.
While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing I could see two variants:
- Dump the traffic
Le 09/04/2012 13:42, Johan Wilfer a écrit :
After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed.
While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.netwrote:
At first, if your Asterisk is in a VM install it on the real server, it
solved us on some installations.
We've gone away from VMs altogether.
To monitor the traffic, you can use voipmonitor.org
We purchased the
2012-04-09 20:22, Carlos Alvarez skrev:
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
At first, if your Asterisk is in a VM install it on the real
server, it solved us on some installations.
We've gone away from VMs
OpenVZ is not really virtualisation, though for some reason people insist on
throwing it into the same discursive space as Xen, VMware, HyperV, etc.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax:
The image you provided didn't open so I'm not sure about the design. If you
can send some SIP flow diagram and Asterisk CLI logs maybe it'll help
understand the problem.
On Fri, Sep 16, 2011 at 1:28 AM, Gilles codecompl...@free.fr wrote:
Hello
My ISP provides an FXS port to plug a
On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com
wrote:
The image you provided didn't open so I'm not sure about the design.
Sorry about that. It's a PNG file and it opens in the two browsers I
tried.
The reason I don't simply get a subscription with a VoIP provider and
must go
The image just don't open for me, a wild from appears and tells me Domain
blocked bla bla. Try attaching image in this mail.
Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at
home connected to their ADSL modem so that they can make free calls
from overseas?
LOL- Its like
Hi Gilles,
Sorry about that. It's a PNG file and it opens in the two browsers I
tried.
It opens here too. It's very simple though. I would put it like this:
VOIP phone ---SIP over the internet--- Asterisk ---internal FXO
card--- PSTN-outlet ---PSTN--- PSTN phone
Can Dahdi/Asterisk do that?
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
I think this is a very common situation, so I'm not really sure what
your problem is. Perhaps it's because I don't use an internal card,
but in my situation it works just fine. I dial a number on my SIP
phone, Asterisk
On 09/16/2011 06:13 AM, Gilles wrote:
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
I think this is a very common situation, so I'm not really sure what
your problem is. Perhaps it's because I don't use an internal card,
but in my situation it works just fine.
On Fri, 16 Sep 2011 10:54:48 -0500, Kevin P. Fleming
kpflem...@digium.com wrote:
This is true, but you already answered your own question in your
original post: since Asterisk cannot know whether the called party
(dialing out via an FXO port) has answered or not, it assumes the
outgoing call is
It does on PRI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, September 16, 2011 7:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Monitoring second leg being
On Fri, 16 Sep 2011 19:35:19 -0400, Eric Wieling ewiel...@nyigc.com
wrote:
It does on PRI.
Unfortunately, this is for an ADSL modem, hence the connection to its
FXS port :-/
--
_
-- Bandwidth and Colocation Provided by
Thanks for the confirmation. Too bad Dahdi doesn't provide
call supervision so that Asterisk knows if/when the callee
has answered.
I'll experiment and see how it goes.
DAHDI with an FXO card can support call answer/hangup supervison.
Check out chan_dahdi.conf options;
Hello
My ISP provides an FXS port to plug a handset, which can be used to
make free calls to (GSM) cellphones, similar to the Billion ADSL
modems:
http://au.billion.com/product/voip.php
My plan is to install an SIP client on a smartphone, so that when I'm
travelling, I can connect to a
On Tue, 12 Jul 2011 11:10:28 -0400, Steven Stromer
fil...@stevenstromer.com wrote:
A quick to implement open source network monitoring tool is smokeping:
http://oss.oetiker.ch/smokeping/index.en.html
Thanks guys for the tip on qualify=yes and SmokePing.
--
On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com
wrote:
Community can help you better if you provide some details about you scenario
and requirement.
It's a very simple scenario: The Asterisk server is connected to a
VoIP provider for calls to the PSTN, and I'd like to have
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Tuesday, July 12, 2011 3:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Monitoring connection to VoIP provider
A quick to implement open source network monitoring tool is smokeping:
http://oss.oetiker.ch/smokeping/index.en.html
Tobi Oetiker and Niko Tyni's awesome tool can monitor latency on a number of
layers, and maintains charted records of connection quality.
It has a probe specific to SIP:
Hello
I was wondering if Asterisk can be configured to monitor a
connection to a VoIP provider, whether someone is currently using it
for a call or the connection is idle?
FWIW, my VoIP provider doesn't run an iperf server on their side. I
don't know if ping/traceroute is a good enough
@lists.digium.com
Subject: [asterisk-users] Monitoring connection to VoIP provider?
Hello
I was wondering if Asterisk can be configured to monitor a
connection to a VoIP provider, whether someone is currently using it for a
call or the connection is idle?
FWIW, my VoIP provider doesn't run
I know that Asterisk can use the system's sound card as the output device
for a console channel. However, I'm using Asterisk call files and would
like to be able to hear the calls over a set of speakers as the call files
are being processed. Basically I'm wanting to listen in on the calls as
PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitoring calls via sound card
I know that Asterisk can use the system's sound card as the output device
for a console channel. However, I'm using Asterisk call files and would
like to be able to hear the calls over a set
Myles Wakeham a écrit :
[...] Are there tools or
add-ons available for this that will email me when a SIP registration
goes offline?
Any suggestions for this would be greatly appreciated.
Hi Myles,
first, best wishes to the list for this new 2010 year.
To answer your question, you
I have an Asterisk 1.4.2 server with 3 different SIP providers and
Asterisk for Skype gateway installed. Periodically the SIP providers go
offline for some reason, or the Skype connection fails.
When this happens, I lose my SIP registration to the provider.
Unfortunately I don't know this has
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server availability.
I'd like to add Asterisk to the servers that our monitoring systems are
handling, and also
, 2009 10:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitoring Asterisk uptime
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server
David writes:
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep
USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework
On 7/08/09 2:28 AM, Myles Wakeham wrote:
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server availability.
I'd like to add Asterisk to the servers that our
I am monitoring a channel... I then redirect that channel to a conf with
lq as options.
When playing back the gsm file I have all recording upto the point of
redirect to
the conference.
How do I CONTINUE to record and not loose anything after redirecting to
the conference?
After redirecting I
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not available messages
when trying
Jerry Geis schrieb:
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not
Jerry Geis wrote:
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not
Hello all -
We are trying to implement some monitoring systems for our production
asterisk boxes. We use whats up gold for all our other stuff. I'd like to be
able to monitor the status of PRI's. For example if a PRI is in alarm, i'd
like to get an e-mail notification. How are others
Hello Jon,
Maybe you can think about SNMP support in Asterisk.
Also you can develop custom applications in many languages or take a look to
Nagios (http://www.nagios.org/)
Try that command on your Asterisk box:
asterisk -rx 'pri show spans', it returns PRI status.
Good lucks
On Wed, Nov 19,
: [asterisk-users] Monitoring
Hello Jon,
Maybe you can think about SNMP support in Asterisk.
Also you can develop custom applications in many languages or take a look to
Nagios (http://www.nagios.org/)
Try that command on your Asterisk box:
asterisk -rx 'pri show spans', it returns PRI status
[EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Sent:* Wednesday, November 19, 2008 7:07 AM
*Subject:* Re: [asterisk-users] Monitoring
Hello Jon,
Maybe you can think about SNMP support in Asterisk.
Also you can develop custom
Thanks!
I'll give this a try
- Original Message -
From: Hakan C
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, November 19, 2008 8:17 AM
Subject: Re: [asterisk-users] Monitoring
Hey Jon,
You are asking something too specific.
If you
ing I could just use?
-
Original Message -
From:
Hakan
C
To:
Asterisk Users Mailing
List - Non-Commercial Discussion
Sent:
Wednesday, November 19, 2008 7:07 AM
Subject:
Re: [asterisk-users] Monitoring
Hello Jon,
Maybe you can think about SNMP support in As
On Wed, 19 Nov 2008 15:17:50 +0200
Hakan C [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Down/', $checkPRI, $match) {
echo OMG,
is this for php?
- Original Message -
From: federico fetto [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 19, 2008 8:41 AM
Subject: Re: [asterisk-users] Monitoring
On Wed, 19 Nov 2008 15:17:50 +0200
Hakan C [EMAIL PROTECTED] wrote:
Hey Jon
Thanks I can work with this.
-Jon
- Original Message -
From: Giorgio Ciccarelli
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, November 19, 2008 8:36 AM
Subject: Re: [asterisk-users] Monitoring
Hello Jon,
you can see in the proc filesystem
Hi,
the command asterisk -rx 'pri show spans' on asterisk 1.2
doesn't work, work only on asterisk 1.4
=-O
federico fetto wrote:
On Wed, 19 Nov 2008 15:17:50 +0200
"Hakan C" [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want to monitor your
fetto [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 19, 2008 8:41 AM
Subject: Re: [asterisk-users] Monitoring
On Wed, 19 Nov 2008 15:17:50 +0200
Hakan C [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want
Hello, everyone,
You Sanem which I can use software to track use of channels of ZAP asterisk.
I have 4 asterisk servers with each 4E1, I would like to monitor the doors of
E1, someone knows a tool for that?
Thank you very much.
Rodrigo
Florianópolis - Brazil
Howdy,
Running asterisk 1.4
Is there a way to check the simultaneous sip calls in asterisk and display with
mrtg or some graphing app???
Also is there a way to segregate these based on extension or context?
Cheers,
Taff..
___
-- Bandwidth
://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of carl Lougher
Sent: Thursday, September 25, 2008 11:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
Hello Fellow Users,
I am looking for a way - using certain software or other techniques - to
monitor, measure, and improve the quality of service for Asterisk system.
During the last while, it seems the quality has decreased and am trying to
look for ways to get things going well again.
Thanks,
Hi,
While I haven't personally used any of their equipment yet, Brix is
supposed to have good h/w and software for measuring a MOS score:
http://www.brixnet.com/products/BrixCall.shtml
http://www.voiptroubleshooter.com/basics/mosr.html
-- James
Hello Fellow Users,
I am looking for a way -
Hi all,
I'm using asterisk to provide a simple service for official time (hour +
minutes + seconds).
The system and application (asterisk + zap detection + custom application) is
monitored by Nagios with some scripts I have created using examples from
voip-info.org.
But I still need to
Hi list,
Recently I figured out how to automatically record (Monitor) both
incoming and outgoing calls, which is handy. However, since this is
not always desirable (or legal), can Asterisk be configured to start
recording at some arbitrary point during a call, to be determined by
the
On 1/21/08, Jaap Winius [EMAIL PROTECTED] wrote:
Hi list,
Recently I figured out how to automatically record (Monitor) both
incoming and outgoing calls, which is handy. However, since this is
not always desirable (or legal), can Asterisk be configured to start
recording at some arbitrary
Hi friends.
I want to monitor my system (Asterisk 1.4.15 with PostgreSql) in real time, I
am using CentOS 5.1 and try with the article in
http://www.voipphreak.ca/archives/382, but I got:
asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_snmp.so:
undefined symbol: init_agent
I need
Hi all,
Is there a way to keep track in Asterisk of which phones are online in
realtime using some MySQL DB table for exemple, much like sip show
peers does in the CLI?
Regards,
Ricardo.
___
--Bandwidth and Colocation provided by Easynews.com --
Ricardo Carvalho wrote:
Is there a way to keep track in Asterisk of which phones are online in
realtime using some MySQL DB table for exemple, much like sip show
peers does in the CLI?
If you are using real realtime with rtupdate=yes in sip.conf
Asterisk stores the current time + sip
Hello list,
we are pleased ro announce that we have released a newer version of
QueueMetrics (1.3.3) that is able to monitor multiple Asterisk servers at
once, thus making it possible to monitor call centers running on clusters
or on high-availability configurations. See
Hi Steve,
Ok Playback could be used here, indeed.
But if you are using automonitor - by default activated by (*1) - I
think there is no way how to implement this.
Am I right?
Thanks,
Ondrej
Steve Totaro wrote:
[EMAIL PROTECTED]
wrote:
Hello,
I'm discovering asterisk, it seem
I think you are right or i didn't find how to to it without using a
conference.
And even with conference didn't find a smart way to make it.
Ondrej Valousek a écrit :
Hi Steve,
Ok Playback could be used here, indeed.
But if you are using automonitor - by default activated by (*1) - I
think
Hi,
During off hours, a server of mine simply forward incoming calls to an
outside number, so that no user is locally available to report or notify
downtimes.
As availability is here a major requirement, I'm looking for a cost
effective and reliable way to monitor this server.
Should I simply
Somewhere did I see a test script.
I will see if I can find it once more.
With that information should you be able to write a simple script that
monitor the server and then will notify you if the server stop responding.
PING wold maybe also be a help.
//Mattias
On 29/11/06, Olivier [EMAIL
Hello,
I'm discovering asterisk, it seem to be a great soft.
I have seen a fonction to record calls that's a great fontion but there is
something disturbing me.
When the record start, except if the recorder prevent the other part, he is not
aware of the recording...
I dont find a way from the
[EMAIL PROTECTED] wrote:
Hello,
I'm discovering asterisk, it seem to be a great soft.
I have seen a fonction to record calls that's a great fontion but there is
something disturbing me.
When the record start, except if the recorder prevent the other part, he is not
aware of the recording...
Hi
I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?
Has anyone done this?
Thanks
SP
___
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asterisk-users mailing list
To
I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
___
--Bandwidth and
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To UNSUBSCRIBE or update options visit:
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Hi,
often on this list I read about transcoding as the heaviest activity for an
Asterisk server, together with high IRQ rate (especially with Digium
cards...).
Is there a way to monitor if Asterisk is engaged (by mistake or by design)
in transcoding or any other heavy activity?
Or a checklist to
I know you can set up monitoring of queued calls, and I'm pretty sure my
question's been answered before, but has anyone devised of a way to
actually barge into a queue channel so you can do in place monitoring of
calls?
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
Yeah, this email took way too long to hit the list. ChanSpy is my friend
:)
On Tue, 23 May 2006, Aaron Daniel wrote:
I know you can set up monitoring of queued calls, and I'm pretty sure my
question's been answered before, but has anyone devised of a way to actually
barge into a queue
As I known, there are many gateway provide SNMP support.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sangoma
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Sent: Friday, May 19, 2006 4:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] monitoring sangoma cards via snmp
Message: 2
Date: Fri, 12 May 2006 09:39:55 +0200 (CEST)
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] monitoring sangoma cards via snmp
To: asterisk-users@lists.digium.com
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Hello,
Digium does not provide snmp
Hello,
Digium does not provide snmp support to monitor their
cards !
Anybody has tried Sangoma product A104 Quad T1/E1 or
others ?
Regards
harry
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Yahoo! Mail réinvente le mail !
[EMAIL PROTECTED] wrote:
Hello,
Digium does not provide snmp support to monitor their
cards !
That's like saying Toyota doesn't provide gas with their cars. You can
setup snmp with in linux and have it execute commands that you want to
determine whether or not the hardware is functioning
Hello all!
Weve been thinking of using the monitor or the
mixmonitor application for a while. However, we have met some basic problems in
getting this to work as planned:
First of all, most of our calls come in through a call
queue. Theres a monitoring option in the queue, and it
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