Try this * 1.4 compilation guide for CentOS:
http://astrecipes.net/index.php?n=286
Thanks
l.
On Sat, 10 Nov 2007 09:12:39 +0100, Mark Quitoriano
[EMAIL PROTECTED] wrote:
Hi im using centos 5 what is the prerequisite to be installed before
compilling asterisk 1.4?
Thanks!
--
Loway
We offer a very comprehensive reporting and real-time monitoring
commercial solution called QueueMetrics that has also a free mode for
smaller CCs and hobbysts and scales well to multi-server setups with
hundreds of live agents. See http://queuemetrics.com
As a completely free alternative,
If you want to do this automatically, what you're looking for is a
(Predictive) Dialler for Asterisk. There are a few available, both on the
commercial and the free side. I'd start by checking out ViciDial /free)
and SineDialer (commercial) that are some of the most used ones.
Thanks
l.
On
Nice job! I took the liberty to post it on AstPligg as well:
http://tinyurl.com/268bac
Thanks
l.
In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith [EMAIL PROTECTED]
ha scritto:
On Mon, 2007-10-22 at 09:39 -0400, end1r wrote:
I’m interested in what software (Free or course) that people
QueueMetrics is able to prepare a realtime screen meant for a video
projector or large LCD screen to display to show call-center stats in
real-time. We have quite a number of customers who used old linux boxes
connected to the right display that just start up, start firefox and go to
a
Try this: http://astrecipes.net/index.php?n=42
l.
On Thu, 18 Oct 2007 06:43:11 +0200, satish patel
[EMAIL PROTECTED] wrote:
I am new in asterisk world can u shortly explian how to create queue and
how to work this ?
David Gomillion [EMAIL PROTECTED] wrote:
On 10/17/07, satish patel
Hello Marco,
could you explain how you did the interfacing to the Asterisk PBX? does
your prototype speak SIP to receive commands?
Thanks
l.
On Thu, 18 Oct 2007 12:27:33 +0200, marcotasto [EMAIL PROTECTED]
wrote:
Hi All,
sorry if I post again this e-mail but I think the first one was
It's not technically complex to do - you can probably use the astdb for
that, or store all incoming numbers with timestamp in MySQL and run
something like:
SELECT count(*) 5 AS blacklisted
FROM incoming_calls
WHERE callerid = 12345
AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE )
you
that is another reason for leaning
towards
MySQL.
Thanks for the nudge toward MySQL.
On 10/18/07, Lenz [EMAIL PROTECTED] wrote:
It's not technically complex to do - you can probably use the astdb for
that, or store all incoming numbers with timestamp in MySQL and run
something like
Well, most of the configuration (but the dialplan) can be kicked up pretty
fast by preparing a block (eg for a SIP extension) and then pasting it
over with minor modifications. I usually keep the original demo config
files around, that already include most options, and use them as a
If what I see actually deployed is what people are happy with, I'd say the
lion's share surely goes to FreePBX. Another GUI that lots of people like
when they try it is Voiceroute Druid, but it's a commercial product. Of
course, no GUI gives you the amount of flexibility that a text editor
.
-Original Message-
From: Lenz [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 16, 2007 12:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
If what I see actually deployed is what people are happy with, I'd say
I think you should use a set of queues - if your skill-based requirements
are the usual suspects (speaking different languages) it's fairly easy to
set up with a master queue for each language with different priority
groups based on how good the agent is with that language. We have a good
Hello Cosmin,
it's hard to tell without first knowing what is going on on your side, but
I would not just drop call files and let Asterisk decide when to process
them - if you have hundreds of faxes pending, you risk having all lines
busy sending faxes and your other users without a dial
Yes - use the manager API to do an Originate by setting variable $CMD to
the shell code you want to execute, and then call a piece of dialplan
where the shellout will be executed through the System( $CMD ) command.
Note that this would enable an attacker to execute arbitrary commands with
Hello John,
we have a number of customers using each of the solutions you mention and
they all seem to be working correctly. Unless you need a very unusual or
extremely large setup, my suggestion is to go for the one that better fits
your problem space / usage needs.
I hope this helps
l.
Maybe I was lucky, but a client of mine has a 24 FXO TDM2400 and works
like a charm :)
l.
On Sun, 07 Oct 2007 03:06:52 +0200, C F [EMAIL PROTECTED] wrote:
Because they tried competing with the channel bank market. But guess
what, it has only one competitive edge, it's cheaper. But if you
I understand you - it's better to settle down for a few hours with a book
of the dead-tree kind. :)
You could also try SIP Beyond VoIP - it's not just on SIP, but it gives
you a broader usage/adoption scenario.
l.
On Mon, 08 Oct 2007 13:42:01 +0200, Justin Case
[EMAIL PROTECTED] wrote:
Mee too, a lot of the messages I'm sending seem to disappear.
l.
In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman
[EMAIL PROTECTED] ha scritto:
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is
I believe that using the Local/[EMAIL PROTECTED] format will give you a bit
more
flexibility in the dialplan design, as there is an added degree of
indirection. In the end I think this is only marginally costier than the
raw channel format (unless you use the /n option) and should provide
This asrticle was meant as a backup, but I guess it's basically the same
thing as what you are looking for: http://astrecipes.net/index.php?n=93
I hope this helps
l.
In data Mon, 01 Oct 2007 20:08:20 +0200, Robert DeVries
[EMAIL PROTECTED] ha scritto:
I am having some hardware problems
Excellent! - posted on
http://oinko.net/astpligg/story.php?title=Asterisk_clusters_with_a_foneBRIDGE2
Thank you
l.
In data Sun, 26 Aug 2007 12:56:04 +0200, Vicente Aguilar
[EMAIL PROTECTED] ha scritto:
Hi
I've published my Asterisk/foneBRIDGE2/heartbeat setup: config files,
scripts...
Well done! It's top-news on AstPligg right now.
http://oinko.net/astpligg/story.php?title=DUNDi_So_Easy_A_Caveman_Could_Do_It
Thanks
l.
On Wed, 22 Aug 2007 03:51:51 +0200, JR Richardson
[EMAIL PROTECTED] wrote:
Here you go folks:
ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf
If someone would
You may want to start from here: http://astrecipes.net/index.php?n=204
l.
On Sun, 19 Aug 2007 00:46:45 +0200, Ade Vickers
[EMAIL PROTECTED] wrote:
Hi...
I have what is, I am sure, a relatively common straightforward problem
(no, NOT that kind of problem!)... I'm trying to hook two
There's a script here that does various conversions... locate the line
converting to gsm: http://astrecipes.net/index.php?n=153
Bye
l.
On Sun, 12 Aug 2007 22:02:06 +0200, Andrew Joakimsen [EMAIL PROTECTED]
wrote:
On 8/12/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote:
have anyone an idea
I see that the TrixBox Pro website is available now:
http://www.trixbox.com/products/trixbox-pro/
From what I'm reading, there is a free version available, plus two other
versions, one at $9.99/per user/per month and the other at $19.99/per
user/per month.
They have a centralized
I have never thought about it, but you may want to have a look at some
http unit-test framework - they usually provide proxy services that are
able to automate and script a generic http conversation.
l.
In data Fri, 10 Aug 2007 12:13:17 +0200, Olivier [EMAIL PROTECTED] ha
scritto:
hello,
Try MixMonitor()
l.
In data Thu, 09 Aug 2007 00:24:47 +0200, Wai Wu [EMAIL PROTECTED] ha
scritto:
Hi all,
Is there a way to have this command to record a mixed file instead of
one for in and one for out? I have set the Mix parameter to 1, but it is
still generating two files. I would
Hello list,
I posted a couple of tutorials lately, maybe someone can benefit from them:
The first tutorial explains how to transform your Asterisk call recordings
(in WAV or GSM) to lo-fi MP3 to save a lot of space. It's actually pretty
easy to implement using a makefile.
I have used this freeware tool in the past:
http://sineapps.com/sinestatiax.php
maybe you can have a look at it as well
l.
In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED] ha
scritto:
At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
At 12:31 PM -0700 2007/8/3,
It depends - I believe mp3 8k has about the (poor) quality of gsm, but
takes about half the disk space yes, I would not routinarily save in
gsm and then turn it to mp3, but I developed a script for some guys who
had a lot of existing gsm files they wanted transcoded :-)
l.
In data
As a different approach, QueueMetrics includes a perl script that does the
real-time uploading of queue_log data into a database. It is being used in
a large number of high load installations worldwide, so I'd say it's a
pretty proven solutions, and it's very lightweight. As an added bonus,
Hello Jay,
you may want to have a look at QueueMetrics - everything you're looking
for is already there. :-)
l.
On Thu, 26 Jul 2007 16:37:56 +0200, Jay Moore [EMAIL PROTECTED]
wrote:
Greetings, list!
My boss would like some statistics on how many calls are answered out of
specific
Hello list,
I have prepared a new tutorial for Astrecipes on how to compile the latest
Asterisk 1.4 with H323 support, Google Talk and Zaptel support, starting
from a stock TrixBox system.
You can find it here: http://www.astrecipes.net/index.php?n=286
I hope somebody will find it useful :-)
It's messages from the list that get delivered after a few days.
l.
In data Fri, 29 Jun 2007 23:31:44 +0200, Mojo with Horan Company, LLC
[EMAIL PROTECTED] ha scritto:
Is it taking a while for _your_ messages to post to the list, or do you
mean messages from the mailing list software take
Hello list,
I am getting the list with days of delay, take for example this message:
Received: from unknown (HELO lists.digium.com) (216.207.245.17) by
mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 -
Received: from localhost ([127.0.0.1] helo=INXS.digium.internal)
Hello list,
After a few months of developement, we are proud to release QueueMetrics
1.4.
This release adds a very large number of new features and bug fixes, for
example:
- New master engine! It should be 4x faster and 2x as memory efficient as
QM 1.3, though it's tracking much more
at 01:42 +0200, lenz wrote:
Hello list,
AstPligg is a new Digg-like website devoted to * and VoIP news.
At the moment, it's in beta stage and very basic - no fancy custom
templates. It allows posting new stories, comments on stories, RSS feeds
and tags. Still, it can be very useful
that the name could be better, but after having
just tried it out, I really like AstPligg.
AR
On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote:
Great! Another one. With such a catchy name too!
On Tue, 2007-06-26 at 01:42 +0200, lenz wrote:
Hello list,
AstPligg
Hello list,
AstPligg is a new Digg-like website devoted to * and VoIP news.
At the moment, it's in beta stage and very basic - no fancy custom
templates. It allows posting new stories, comments on stories, RSS feeds
and tags. Still, it can be very useful, as the number of * sites and blogs
Hi Lee,
we are a Java shop and our experience with Java has been much the one you
say - it does scale pretty well and it is very solid. What I was trying
to say is that Java is not very well suited to the classic, Unix-style,
fire-up-process-and-let-it-die that goes for CGI/AGI
Hello Matthew,
Java is not a great solution for AGIs because they are script you should
fire up and terminate very fast, while the overhead of launching a JVM,
loading all classes, etc, is pretty large. Also, you don't want multiple
JVMs in parallel loading everything multiple times.
You may also want to have a look at our suite QueueMetrics, that is
deployed in hundreds of CCs worldwide, is very flexible and is free for
small CCs. See http://queuemetrics.com
I hope this helps
l.
On Fri, 25 May 2007 02:02:18 +0200, Senad Jordanovic [EMAIL PROTECTED]
wrote:
bilal
Hello Erick,
I believe that if you go for a manual installation of non-AsteriskNOW
components (like Java) they should be excluded from the components that
Conary mantains.
l.
On Mon, 21 May 2007 09:54:52 +0200, Erick Perez [EMAIL PROTECTED] wrote:
I realized that queuemetrics uses
scritto:
lenz wrote:
Is the queue enidan configured at all in queues.conf? and how is it
defined?
l.
Sorry, I should have added that:
from queues.conf:
[enidan]
strategy = ringall
;announce = enidan-queue
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
member = Local
Hi Chong,
I have no experience with MGCP, but do you see anything in the Asterisk
CLI or the full log while the terminal is supposedly being called by the
ACD?
Another thing you could do is to create a Local/[EMAIL PROTECTED] channel, whee
as the queue memebre you have Local/[EMAIL
Is the queue enidan configured at all in queues.conf? and how is it
defined?
l.
In data Mon, 14 May 2007 13:56:25 +0200, Per Jessen [EMAIL PROTECTED] ha
scritto:
I have a queue defined which I use like this:
exten = _X.(reception),n,Ringing
exten = _X.,n,Queue(enidan,t,,,20)
exten =
It sounds pretty interesting to do. On a technical point of view, I
believe a simple HTTP interface would be nice, and additionally a nightly
rsync for high load sites who wish to run a local cache.
About how to set up a community blacklist for telephone numbers and its
legal implications,
Not sure about the rest of the world, but here in Italy virtyually all
telemarketers will call using a blank caller-id, so you have no way to
know who they are and/or call back.
Spoofing might be a problem, though I believe it will take a bit of users
for the blacklist to be on
Why don't you make up the MOH in order to play your sound files, as you
need?
l.
On Mon, 07 May 2007 16:29:28 +0200, Andre Courchesne - Consultant
[EMAIL PROTECTED] wrote:
Hi,
Anyone knows if there is a way to play a list of sound file in a
round robin mode (at specific interval)
Hello list,
hope someone can help me - I'm going crazy using the FXO port a SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that
is, once it detects a call, it should simply send it over to the local
Asterisk server. No intelligent routing, PIN, anything
Thanks a lot, that did the trick! Wish there was an half-decent manual on
the site at least :-(
l.
In data Thu, 03 May 2007 18:34:34 +0200, Dave Cotton
[EMAIL PROTECTED] ha scritto:
On Thu, 2007-05-03 at 17:56 +0200, lenz wrote:
Hello list,
hope someone can help me - I'm going
Maybe you have too short a digit timeout.
l.
In data Sun, 22 Apr 2007 11:39:59 +0200, Poul Moller
[EMAIL PROTECTED] ha scritto:
Getting better... however still l didn't managed to transfer a call from
my
ATA. As you see some digits new gets recognized but never the full
extension
(1002
Why don't you simply pause them when they are unavailable?
l.
In data Thu, 19 Apr 2007 14:33:52 +0200, Arun Kumar [EMAIL PROTECTED]
ha scritto:
Hi
I've configured the queue on my asterisk box and everything is working
fine.
In my queue I've 3 agents logged in the queue. When call comes
DumpChan (it's there in 1.2 as well) would be great, if it were a manager
command where you can choose the channel to dump and not a diaplan
function that outputs the current channel config to the CLI.
l.
In data Wed, 18 Apr 2007 02:30:09 +0200, Philipp von Klitzing
[EMAIL
Well, the larger the better :)
l.
In data Wed, 18 Apr 2007 04:15:28 +0200, Melcon Moraes
[EMAIL PROTECTED] ha scritto:
How large is large for you?
[]'s
MM
-Original Message-
From: Lenz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello list,
QueueMetrics 1.3.4 has been released today. Among other features, it
provides realtime cluster monitoring through the manager API and, by
popular demand, user defined time intervals in the daily call breakdown.
You can find the latest version at http://queuemetrics.com and
Hello list,
we are developing a new application that uses the Manager API in order to
find a set of channels where variables are set in a predefined way. To do
this, we currently send a Status command to obtain all available channels
and then query them all, one by one, fot the status of a
Hello list,
I have been working lately on a small CDR parsing utility, and would like
to do some performance testing on it. I am looking for some - possibly
large - real-life Asterisk CDR datasets to run some performance
monitoring. Anybody's got some CDRs that can be shared?
Thanks in
.
On 4/17/07, Lenz [EMAIL PROTECTED] wrote:
Hello list,
I have been working lately on a small CDR parsing utility, and would
like
to do some performance testing on it. I am looking for some - possibly
large - real-life Asterisk CDR datasets to run some performance
monitoring. Anybody's got some
it :)
Thanks for your offer anyway - I'll check it!
l.
In data Wed, 18 Apr 2007 01:09:54 +0200, Earl Terwilliger [EMAIL PROTECTED]
ha scritto:
Hi Lenz,
Why not do it the same way as you do the queue log (for queuemetrics)?
i.e. have a program which captures all events (or certain events
Hello Eric,
if you are looking for queue_log or CDR data parsing, there are a few
tools out there that can do it. If you are looking to parse Asterisk's own
debug activity log (like full / messages) you should likely look for a
general unix log parse tool.
l.
In data Wed, 21 Mar 2007
If forwarding happens after a number of rings, you could simply cut off
the call before the required number of rings happen. Otherwise, I' don't
think there is a simple way to detect being answered by the voicemail
versus the intended recipient.
l.
On Wed, 21 Mar 2007 13:23:37 +0100,
Hello list,
we noticed that in some conditions Asterisk would not log queue exit
records, thus producing a queue_log lacking some end-of-queue events. I
have then prepared a small tutorial to get you started on tracking queue
exit conditions that cause the problem.
Here it is (I hope thi is the one Steve was speaking of!) :)
http://groups.yahoo.com/group/astcallcenters/
Hope this helps
l.
In data Thu, 15 Mar 2007 21:35:06 +0100, nik600 [EMAIL PROTECTED] ha
scritto:
i haven't found any call center asterisk mailing list, but i've found
this:
You could set a dialplan variable in the AGI so that it's pretty easy to
tell what happened in the AGI.
About the code 0, the funny part is that you see AGI Script completed,
returning 0 even if the AGI does not exist, or is not executable. This
should be a good candidate for improvement
Hello list,
I have prepared a small tutorial today that deals with how to avoid
Asterisk rebuilding DTMF tones when using it to connect industial
appliances that use DTMF. You can find it at:
http://astrecipes.net/index.php?n=248
I know it isn't everybody's piece of cake, but I thought
Not sure about * 1.4, but you can definitely use our Qloaderd script to do
that - see http://queuemetrics.com/download.jsp . That script is pretty
smart (to be a loader script...) and is able to handle restarts and
database disconnections.
l.
In data Thu, 22 Feb 2007 09:20:59 +0100,
Well, kind of - it is meant for weird situations where mostly you do not
have regular POTS phones. Of course all DTMF detection would be disrupted.
l.
In data Thu, 22 Feb 2007 18:59:50 +0100, Yuan LIU [EMAIL PROTECTED] ha
scritto:
From: lenz [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 17:30
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
That's what I get:
The requested URL / was not found on this server
:)
l.
On Mon, 29 Jan 2007 23:16:47 +0100, Facundo Ameal [EMAIL PROTECTED] wrote:
Hello everyone! In my humble try of creating a Blog, I've made this:
http://fameal.blogdns.org.
By now, it's hosted in my own server but
Kind of - you could link that to the Local/xxx channel called for agents,
or you could fork the dialplan and on one branch send the user to the
queue and on the other one run the AGI.
l.
On Mon, 29 Jan 2007 15:55:21 +0100, nik600 [EMAIL PROTECTED] wrote:
Hi everyone
dou you know if is
Did you try Druid from Voiceroute? it's a commercial product, but we find
it pretty handy.
l.
On Mon, 29 Jan 2007 04:09:38 +0100, Santiago del Castillo
[EMAIL PROTECTED] wrote:
Hi, I'm looking a queue manager compatible with queues.conf. It should
allow me to change agents from one
Yes, I confirm the autofill option is present in 1.4, but must be enabled
manually not to break compatibility with 1.2.
l.
On Fri, 19 Jan 2007 15:32:32 +0100, Tom Rymes
[EMAIL PROTECTED] wrote:
You may be running into the limitation in Asterisk 1.2 (It's fixed in
1.4, I think
What you have been told is correct, seems like there is something strange
in your setup then, with * not logging correctly. I'd try running a couple
more calls to see if the problem persists.
l.
On Thu, 18 Jan 2007 23:03:28 +0100, Danny Lan M. - Telegroup®
[EMAIL PROTECTED] wrote:
Hello Chris,
we have a number of clients who deployed very large CCs over the 200 agent
range.
Your idea #1 is pretty sound and I believe that's what most people are
doing. I would like to add a couple of points of attention:
- having hundreds of agents on a box means a lot of synchronous
I implemented something on these lines for unattended transfer. Basically
what I did was storing the call-id in an inheritable diaplan variable and
then starting a new mixmonitor on the transferred extension.
Hope this helps,
l.
On Mon, 15 Jan 2007 22:45:49 +0100, Jay Moore [EMAIL
Hello Yuan,
I have recentky spoken to a number of customers who run call-centers,
tried 1.4 test installs and concluded it's not there yet in terms of
reliability. If I were to install a production box today, I would go for
1.2.
l.
In data Mon, 15 Jan 2007 00:01:27 +0100, Yuan LIU [EMAIL
I have not yet seen this article posted to this list, so I thought many of
us would be interested in having a look at this project sponsored by
O'Reilly:
http://www.oreillynet.com/etel/blog/2007/01/help_create_the_asterisk_cookb.html
It seems they are looking for both problems and
We offer a commecial very detailed reporting solution that is widely
deployed and is available free of charge to small CCs / SOHOs. See
http://queuemetrics.com . Which kind of call center are you going to
implement? inbound / outbound / mixed traffic?
l.
On Mon, 15 Jan 2007 19:36:11
The easy answer is to use the r switch with Queue, still you may want to
use a MOH that fakes a ringing tone in order to have audio messages
smoothly mixed in :)
l.
On Thu, 11 Jan 2007 23:33:40 +0100, Ex Vitorino [EMAIL PROTECTED]
wrote:
Hello List,
This must be an easy one...
Hi Jan,
You should use the logrotate in order to delete the log on periodic
intervals. This article is meant to do exactly the opposite :)
http://astrecipes.net/index.php?n=205 but you get an idea of how to setup
log file rotation and how to notify Asterisk that it should open a new
file
without having the script? What will the command do?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Lenz
Skickat: den 8 januari 2007 13:13
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Manage 'full' log
I think we are going to do it if we get big problems with those many
queues. From what I'm seeing, the biggest problems seem to be related to
agents, so maybe we can have a try at using straight terminals instead of
agents.
l.
On Fri, 05 Jan 2007 01:14:08 +0100, Leo Ann Boon [EMAIL
, lenz wrote:
Hello list,
one of our clients is going to be deploying a system with over 200
differently composed queues and 100 agents. We are going to do a full
test
of the viability of this solution before deployment, but I was wondering
if anyone has experience of such a setup
You are correct, this is more or less the scenario involved - the problem
is that people want to call a personalized line AND speak to the same
subset of agents preferably.
I have never seen such a setup myself - I have seen CCs with 30 or 40
queues, never 200 - so I was wondering if
differences. Your sript looks very useful thoiugh! :)
l.
In data Thu, 04 Jan 2007 11:13:23 +0100, Gavin Hamill [EMAIL PROTECTED]
ha scritto:
On Thu, 04 Jan 2007 11:05:38 +0100
Lenz [EMAIL PROTECTED] wrote:
You are correct, this is more or less the scenario involved - the
problem
Hello list,
one of our clients is going to be deploying a system with over 200
differently composed queues and 100 agents. We are going to do a full test
of the viability of this solution before deployment, but I was wondering
if anyone has experience of such a setup and if there are any
I would post it to some site of yours (or Sourceforge if you plan to have
shared CVS) plus a page on the wiki, so people can find it. I have been
working on a few projects on sourceforge and never had problems with it.
With licence, you choose. GPL is usually a good starting point for
Are you sure you want to fire up a JVM each and every time you run this
command? that's a resource hog and will anyway cause a delay for system
class loading, etc. Maybe attaching to a resident process would be lighter.
k,
On Fri, 22 Dec 2006 13:48:27 +0100, Andre Gustavo Lomonaco
I have been speaking privately to a number of CC integrators and resellers
about the AgentCallbackLogin() deprecation issue, and I'd dare say nobody
is enthusiastic about it. With all its problems, AgentCallBackLogin is the
workhorse of most of today's Asterisk CCs, and my impression is
I would not say that * has been taken out of the realm of reasonable CC
solutions, luckily there are still a lot of ways to configure * to meet
the most diverse needs, but it's surely a fact that a very convenient and
used approach being deprecated will be an annoyance for CC designers and
it? Convoluted dialplans? AEL stuff?
Agentcallbacklogin was
simple and easy. Hardly anything in Asterisk is simple or easy. Maybe
that's
why it was removed. :)
On Wednesday 20 December 2006 2:21 pm, Lenz wrote:
I would not say that * has been taken out of the realm of reasonable CC
solutions, luckily
You may want to have a look here: http://astrecipes.net/index.php?n=42
Best regards
l.
On Wed, 13 Dec 2006 16:15:17 +0100, Jay Moore [EMAIL PROTECTED]
wrote:
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm
We have a few clients running large CCs who are using them and seem quite
happy with them.
l.
On Tue, 12 Dec 2006 16:21:36 +0100, Cory Andrews [EMAIL PROTECTED]
wrote:
Looking for info recommendations for SIP load balancing, thanks in
advance!
Cory Andrews
--
Loway Research - Home
One thing you could do is use a third-party product like our QueueMetrics
(available free for smaller systems/SOHOs) and use its own internal logic
to link a callerid to all other information (call status, agent, time,
etc), search by different criteria and remote call listening.
Hope
Have you tried putting a Local channel for the dynamic agent and, after
the dial() tries for 20 seconds, you perform a dynamic agent logoff. Not
sure if this will cause deadlocks, removing a dynnamic agent while he's
being called, but maybe worth trying.
Just my two (euro)cents,
l.
That's pretty easy and included in the basic * implementation - you tell
the queue not to accept users and play a message after the queue command
terminates.
l.
On Thu, 16 Nov 2006 09:10:46 +0100, nik600 [EMAIL PROTECTED] wrote:
Hi
i have to manage a particular implementation for a
Hello list,
I am struck with a problem: I have a setup where I accept calls on an FXO
port from the PSTN and redirect them immediately to an FXS port where a
data acquisition appartaus is waiting for data sent in DTMF format. The
problem I am experiencing is that Asterisk will in any case
Hmmm accesisng the GUI using the browser?
Seriously: did you encounter any problems?
l.
On Sat, 11 Nov 2006 12:52:49 +0100, Thirumal Saminathan
[EMAIL PROTECTED] wrote:
hi,
i got following message
please tell me what can i do after this ..
linux:~/asterisk-gui # make checkconfig
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