Hello All,
Iam trying to achive a simple load balancing with dundi.
Here i have three asterisk boxes like below.
*.*.*.121 which is the dundi server
*.*.*.137 A Peer which has the 1000 phone registerd to it
*.*.*.204 B Peer which has the 200 phone registered to it.
The expected behavior of
Dear Tilghman,
Thanks for your feedback. Scratching my head to see how to turn the
standalone box to something more scalable. func_odbc , mysql in
dialplan, all are missing the states of devices, agents... etc
Realtime, as far as it goes, also touch only one part of asterisk, by
storing all the
G'day listers,
Well, I just found this one:
http://svn.digium.com/view/asterisk/team/russell/events/doc/distributed_devstate.txt?view=markup
Time to read.
On Tue, Sep 9, 2008 at 2:29 PM, Nguyen [EMAIL PROTECTED] wrote:
Dear Tilghman,
Thanks for your feedback. Scratching my head to see how
Hello!
I just wonder, I've uncommented a few part of my features.conf, to park
calls and the like. But now I wonder, how it can be done? I tried the #72
(park call) from both asterisk with
CLI misdn send digit mISDN/1-102 #72
Never mind if the channel-name looks wrong here, asterisk
Dear All,
I would like to ask please about how to fix the problem of sending fake ring
back tone by asteriskserver when trying to make a call from an extension
registered on asterisk to any PSTN number...I made some comparaison between
calls made through Asterisk server that generate a fake ring
Paul,Thank you very much for your reply!Recordings
and voicemail are not even the most important thing really, but call
forwarding is. ARI seemed to have all of them mungled in, so I
mentioned it.However, if you know of something that will
require me to add a few contexts to the dialplan and put a
On Monday 08 September 2008 14:44, Atis Lezdins wrote:
On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED]
wrote:
I dont have problem to make a reload by AMI.
My questions was if module reload app_queue.so is the right way to do
this, because whis reload I reload everything.
Dear asterisk-users@lists.digium.com, Best Price 79% 0FF
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On Tue, Sep 9, 2008 at 1:22 PM, Thomas Winter [EMAIL PROTECTED] wrote:
On Monday 08 September 2008 14:44, Atis Lezdins wrote:
On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED]
wrote:
I dont have problem to make a reload by AMI.
My questions was if module reload app_queue.so
Dear All,
I have configured here Asterisk-stat
(Call Detail Records) for CDR ANALYSER. Here I am facing problem
in web analyser when I Selection of the day as I require
it can get data from asterisk postgres database. But in bottom side I
have not seen graphical chart and I also can't
Dear All,
I have configured here Asterisk-stat
(Call Detail Records) for CDR ANALYSER. Here I am facing problem
in web analyser when I Selection of the day as I require
it can get data from asterisk postgres database. But in bottom side I
have not seen graphical chart and I also can't
Hi All,
i have 3 asterisk server in a cluster using a cluster of mysql server
via realtime, users can register via DNS SRV.
I send/receive calls to an AS5400 via a SIP trunk defined on the
realtime sip table, the trunk has call-limit=5. Problem i encountered is
each of the 3 asterisk
Hi
What i need to do:
exten = 1001,1,AGI(Agent.agi)
agent.agi - login my interface in system
i would call to 1001 using Manager API and login interface in Asterisk.
This is possible?
Now i use originate. Something like that:
Action: originate
Channel: SIP/ekiga
Context: default
Exten:
Maybe this will work for you, but I am not sure what files are touched
by ARI. It seems that it would have to touch extensions.conf for
forwarding to work as well as recording (or at least try to write to a
DB). Actually, the more I think about it the less I think it will
work for you, but
Dear Asterisk Users
I'm looking for a solution that can be used to monitor Asterisk and the
Telco lines aswell as the network (Servers, WAN LAN links, Router
Switches)
Thanks
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On Tue, Sep 9, 2008 at 3:19 PM, bala krishnan [EMAIL PROTECTED] wrote:
Hi,
The problem is, when i was starting the recording on zap channels
through AMI by Monitor command, always the out stream recorded as 0 bytes.
So that i did the searching and got the response that t option would
Do a 'make menuselect' when you install zaptel driver and make sure in
the 'Binary Firmware Package' either ' FIRMWARE-OCT6114-064' or '
FIRMWARE-OCT6114-128' is selected depending on which echo cancellation
module you have. If you are unsure select both.
Remi
Klaverstyn, David C wrote:
On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote:
Dear Asterisk Users
I'm looking for a solution that can be used to monitor Asterisk and the
Telco lines aswell as the network (Servers, WAN LAN links, Router
Switches)
We use nagios for that.
--
Michiel van Baak
[EMAIL PROTECTED]
Hi all,
I noticed a strange X-Lite behavior, it's connected to an asterisk box.
The client registers normally and everything works fine. When I dial out
(via E1-PRI) and the called party is unavailable, and asterisk indicates
CONGESTION to X-Lite. So far so good.
When I try to make another
If you have php installed this script can get you started. You can either make
some additions (send an email) and cron it or leave it as and point an external
monitor at the web page doing a content check.
The output if everything is OK is as follows:
Zap Lines OK
If there is an issue:
Issue
Is the idea to switch to another video source or stay with the callers camera? An option for both would be nice. I could see a help desk placing a caller in que and a 1-2 min video coming on showing some simple video of "how to hook it up".
-- Original message from Russell
Has anyone ever 'released' an Asterisk module that is easily
shared/downloadable?
Or doesn't the nagios open source code work like that?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent: Tuesday, 9 September 2008
Nagios has a plugins and a plugins-extra/contrib section and I've seen
*lots* of Asterisk plugins/checkers. As always, consult the Google and
find it -- http://www.google.com/search?q=check_asterisk -- and
Voip-Info also has a page on Nagios check scripts for Asterisk at
Dean,
I'm using Zabbix to monitor network interfaces, storage, cpu load and a
few other things on several asterisk boxes. I'm just looking at adding
Asterisk specific monitoring. Simple things like sip registration is
pretty easy. Getting the actual status of zap-daddy hardware might be a
http://www.voip-info.org/wiki/view/Asterisk+monitoring
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asterisk-users mailing list
To
On Tue, Sep 9, 2008 at 9:42 AM, Paul Schewietzek [EMAIL PROTECTED] wrote:
Hi all,
I noticed a strange X-Lite behavior, it's connected to an asterisk box.
The client registers normally and everything works fine. When I dial out
(via E1-PRI) and the called party is unavailable, and asterisk
On Tue, Sep 09, 2008 at 09:21:50AM -0500, Darrick Hartman (lists) wrote:
I'm using Zabbix to monitor network interfaces, storage, cpu load and a
few other things on several asterisk boxes. I'm just looking at adding
Asterisk specific monitoring. Simple things like sip registration is
On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson [EMAIL PROTECTED] wrote:
I'm faced with an installation at a client site with supposed PRI service on
a fractional T1.
Steve Totaro wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turned up channels go
Asterisk Users -
Just a reminder that we're now only two weeks away from the kick-off
of AstriCon 2008. This year's show is looking really good: at this
point we're expecting over 700 Asterisk users, developers, and
resellers. There are currently 60 presentations scheduled, including
keynotes
Steve,
Thank you for that link!!
However, you saying that it might not work scares me already.. :S
I guess I’ll have to somehow try it out. It would be nice where a the install
needs a block of code pasted into extensions.conf, and a block placed in
/var/www/ and we’re good to go, lol.
Hi All
I am a premium IVR content service provider thats runs on premium rate lines,
my setup (currently on PRIs) is like customer dials the short code (premium
number) which gets forwarded on the PRIs to my IVR. In the normal world the
customer is charged immediately the call is answered by
A simple AGI script would be able to handle that easily, I would think.
Or am I missing something in the details?
-josiah
Sriram wrote:
Hi All
I am a premium IVR content service provider thats runs on premium rate
lines, my setup (currently on PRIs) is like customer dials the short
Hello!
I wondered could I (mis)use an AGI program to decide if I pickup. At the
moment asterisk has to pck up, when the ring tone has stopped playing.
The dialplan looks like this:
*** CUT ***
exten = NUM,1,System(mplayer file /dev/null)
exten = NUM,n,Answer()
exten =
On Tue, Sep 09, 2008 at 11:13:45AM -0400, Bill Michaelson wrote:
On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson [EMAIL PROTECTED] wrote:
I'm faced with an installation at a client site with supposed PRI
service on
a fractional T1.
Steve Totaro wrote:
(Please quote properly next time)
-- next part --
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Message: 2
Date: Tue, 9 Sep 2008 10:14:16 -0400
From: Dean Collins [EMAIL
On Tue, Sep 09, 2008 at 10:55:14PM +0530, Sriram wrote:
I think i wasnt clear here - It'll be either a premium rate line/toll free
line but the customer should be charged Rs.6/- per minute only when he hears
a prompt(where it'll ask him to press 1 to continue) once he presses 1 to
accept
Hello everyone,
We had one of our PBXs crash due to a hardware failure, and rebuilt it with PBX
in a Flash. We are using the current versions of libpri, zaptel and *. It's
the same server with replacement hard drives - a Dell 2850 with a TE410 T1
card, single PRI. It was running v1.2 for
On Tue, Sep 09, 2008 at 10:45:59AM -0400, Jay R. Ashworth wrote:
On Tue, Sep 09, 2008 at 09:21:50AM -0500, Darrick Hartman (lists) wrote:
I'm using Zabbix to monitor network interfaces, storage, cpu load and a
few other things on several asterisk boxes. I'm just looking at adding
Asterisk users,
Voiceroute is proud to announce DruidCON to be held from 1-2 Oct 2008 in
Atlanta GA so that Asterisk users can join us after Astricon :) DruidCON is
the premier meeting for Druid community users and developers with interest
in Unified Communications. DruidCON 2008 attendees will
Dean Collins wrote:
Has anyone ever 'released' an Asterisk module that is easily
shared/downloadable?
Or doesn't the nagios open source code work like that?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent:
I notice that I have only format_ilbc.so but not codec_ilbc.so... is
it due to the compilation or there is some way to create the module?
On Sun, Sep 7, 2008 at 11:14 PM, Stelios Koroneos
[EMAIL PROTECTED] wrote:
Make sure there is no noload = codec_ilbc.so in the module folder
You can also
Hi all!
I am looking for some software that would work as a proxy between a SIP
device (SIP phones and ATA boxes) and the Asterisk system running IAX. The
reason is that I can only get IAX trow the firewalls, and can't change the
settings.
One solution I am using are getting several Asterisk
I would suggest using OpenSIPS with Asterisk and bypass IAX all
together for this particular application.
An OpenSIPS solution will take care of your traveler's NAT issues (and
could handle the registrations) while you used Asterisk for voicemail
and whatever else.
I've personally used
- Original Message -
From: Edgar Guadamuz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 9, 2008 2:18:28 PM GMT -05:00 Columbia
Subject: Re: [asterisk-users] iLBC and G729 codecs
I notice that I have
On Tuesday 09 September 2008 14:00:00 Ming Yong wrote:
Voiceroute is proud to announce DruidCON to be held from 1-2 Oct 2008 in
Atlanta GA so that Asterisk users can join us after Astricon :) DruidCON is
the premier meeting for Druid community users and developers with interest
in Unified
The Asterisk.org development team has released Asterisk versions
1.4.22-rc5 and 1.6.0-rc6, as well as Zaptel version 1.4.12.1. These
releases are available for immediate download from
http://downloads.digium.com/.
This update for Zaptel includes a fix for an issue that would cause the
build to
Lol where have you been the last few months.
There have been at least 4 announcements with dates etc, this is really
just the last chance reminder email.
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 (New York)
+61-2-9016-5642 (Sydney)
http://www.Cognation.net
Hi All,from what i'm understanding, Asterisk is back to back user agent.
Base on this my initial thought was even if we enable reinvite in sip.conf,
asterisk still will be in sip path after transfer.
But i read some information in asterisk using refer to transfer a
call completely to another sip
On Tuesday 09 September 2008 16:36:37 Dean Collins wrote:
There have been at least 4 announcements with dates etc, this is really
just the last chance reminder email.
It's the first I've seen of it. In any case, if this was the last in a series
of reminders, I'm puzzled why the email started
Hi Asterisk users!
I have a little problem with an Asterisk 1.4.22 installation for a
customer. The PBX is connected to an E1 line and we have a few snom 300
attached to it.
The goal is to emulate traditional german PBX behaviour wich is the play
a stuttered internal dialtone after pickup and
Since I play a 70 balance druid on WoW I thought it was something else.
On Tue, Sep 9, 2008 at 2:56 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 09 September 2008 16:36:37 Dean Collins wrote:
There have been at least 4 announcements with dates etc, this is really
just the
Hi,
I just set up my first Asterisk with MeetMe conference support on my local
LAN.
It works great, but I think it may need a little tuning - I am getting audio
delays of up to 1 second. Should I expect better performance in this area,
or is this to be expected?
Thanx!
You shouldn't have any delays at all.
Are you using ztdummy for timing? and what kind of load does the box
have on it?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 9, 2008, at 4:23 PM, George
Darn shame it could not be on a weekend. I would have liked to go.Ronny Julian
-- Original message from Tilghman Lesher [EMAIL PROTECTED]: --
On Tuesday 09 September 2008 16:36:37 Dean Collins wrote:
There have been at least 4 announcements with dates etc,
Hi,
Does anyone use the nagios plugin for check_sip against asterisk?
Does anyone have a working example of the command definition and
service definition in the nagios config files?
TIA
Robert
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PaulH
Jacobus van Niekerk wrote:
Dear Asterisk Users
I'm looking for a solution that can be used to monitor Asterisk and the
Telco lines aswell as the network (Servers, WAN LAN links, Router
Switches)
Thanks
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I have used both munin and nagios - both are cool.
PaulH
EdPimentl wrote:
http://www.voip-info.org/wiki/view/Asterisk+monitoring
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Hello,
please read bellow:
On Tue, Sep 9, 2008 at 11:04 PM, Christian Victor
[EMAIL PROTECTED] wrote:
Hi Asterisk users!
I have a little problem with an Asterisk 1.4.22 installation for a
customer. The PBX is connected to an E1 line and we have a few snom 300
attached to it.
The goal is
Hey folks,
I'm looking to potentially take some of my Asterisk servers and see
how well they fare in a cloud computing environment such as Amazon EC2
+ S3. I was curious to hear feedback from anyone who's willing to
share their experience if they've already done the same. Have you had
a positive
Hi Hiren,
Can you please confirm the php-gd is properly installed?
Thanks,
Max Alex
Voip Developer
On Tue, Sep 9, 2008 at 4:20 PM, Hiren Mistry
[EMAIL PROTECTED]wrote:
Dear All,
I have configured here Asterisk-stat (Call Detail Records)for
CDR ANALYSER. Here I am facing
Asterisk is a well-grounded system service. It does not belong in the
clouds; the weather alone up there is quite pernicious.
Steve Finkelstein wrote:
Hey folks,
I'm looking to potentially take some of my Asterisk servers and see
how well they fare in a cloud computing environment such
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