Hello list,
I have prepared a couple of new tutorials you may find interesting:
- Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216
- Installing the Digium's Asterisk GUI for 1.4 - at
http://astrecipes.net/?n=217
It's nothing too complex, but you may find them
Could it be you did not bind it correctly in http.conf? Something similar
happened to me today while I was doing the same thing.
Try:
enabled=yes
enablestatic=yes
bindaddr=0.0.0.0
Hope this helps
l.
On Thu, 09 Nov 2006 19:28:17 +0100, Curt Shaffer [EMAIL PROTECTED]
wrote:
I was just
On Thu, 26 Oct 2006 07:37:40 +0200, Rajkumar S
[EMAIL PROTECTED] wrote:
Hi Lenz,
On 10/25/06, Lenz [EMAIL PROTECTED] wrote:
if you use Local channels for agents (or callback agents), you can
easily
do this in the Dial() command after the Local channel is called.
I am using call back
why not using a zap show command and parse the results externally?
l.
On Thu, 26 Oct 2006 13:12:46 +0200, Nick Adams [EMAIL PROTECTED] wrote:
I need to determine the number of active calls in a group from outside
of Asterisk. Currently I poll the manager API and parse the channel
status
Hi Raj,
if you use Local channels for agents (or callback agents), you can easily
do this in the Dial() command after the Local channel is called. Of course
your clients may get a bit angry at being disconnected, it is usually
better to jave each agent stay aware od the call length and
Hello list,
I am encountering a bit of a problem in working with incoming calls with a
TDM2400 and * 1.2.4; when a call comes in, * will correctly detect the
ringing, but will sometimes report multiple Attempting native bridge.
What I do is basically that when a call comes in, I dial a
Hi list,
I have a client with a strange requirement: putting a noise gate on the
Asterisk channel. For those who are not familiar with them, noise gates
are used in musical instruments to avoid entering low-level noise into the
amp system. What they basically do is, they measure the volume of the
Hello list,
I am pleased to tell you that we have released a new version of
QueueMetrics. The main areas of improvement were the following ones:
- Internationalization engine: the text of QueueMetrics can be easily
localized, as well as numbers and dates.
- Internationalized versions:
Hi list,
out of pure frustration I have prepared another tutorial (must be the
season) about how to filter the various outputs of Asterisk in order to
keep track of what is going on in realtime in a call-center, to avoid
being swamped by too many logging and information on the * side.
Hi Michael,
do you want to do the reporting or to configure the dialplan? QueueMetrics
will do the reporting for no matter how many ACD queues, and will
automatically sync to the underlying * config files, so there should be no
problem with reporting. You can also configure it in
If you are using queues, you may want to test the free version of
QueueMetrics. You can pass anything to it through the dialplan, and you
can use any phone you want as it will take care of the URL opening on the
client side.
Yours
l.
On Fri, 06 Oct 2006 19:10:27 +0200, Michael Sampson
Hi list,
I must be in tutorial writing mode this week, as I have prepared another
tutorial on how to avoid queue_log file rotation on AAH/TrixBox and other
systems. This is done automatically but it's quite an annoyance because it
interferes with queue_log analyzers like QueueMetrics and
Hi list,
I must be in tutorial writing mode this week, as I have prepared another
tutorial on how to avoid queue_log file rotation on AAH/TrixBox and other
systems. This is done automatically but it's quite an annoyance because it
interferes with queue_log analyzers like QueueMetrics and ends up
by step guide to DUNDi? Good luck with that
though because base DUNDi docs are rarer than periodic element #114 in
the
known universe.
Doug.
-Original Message-
From: lenz [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 11:11 AM
To: asterisk-users@lists.digium.com
I was thinking about that, but there does not seem to be so much interest
in DUNDi at the moment - most people I see are still trying to understand
what a context is and why they cannot use the transfer button in a
queue. :)
l.
In data Wed, 04 Oct 2006 20:01:20 +0200, Douglas
Hmmm... how is your IAX conf between the two boxes B and C?
l.
On Thu, 05 Oct 2006 20:55:51 +0200, Matt [EMAIL PROTECTED] wrote:
Hi,
I am purchasing minutes (800) from provider a (from now on A). My
server is B, and my customer is C. When an 800 call comes in it goes:
A---sip--B--iax--C
Hi list,
today I have been teaching a class on * and have found that many students
find it quite hard to understand how setting up IAX peering between two
servers may work. So I prepared a little step by step tutorial hoping it
might be useful to someone in the future.
See it at
it on you priority 8 in your dialplan
exten = _0.,7,Dial(Zap/g1/${EXTEN:1})
exten = _0.,8,Goto(s-${DIALSTATUS},1)
This is just an example.
Hope it helps, please give me some feeback.
On 10/1/06, Lenz [EMAIL PROTECTED] wrote:
I don't think that is the case - if I add a wait(10) after
Hello list,
I have been puzzled by a behaviour I cannot grasp - the system is an
Asterisk 1.2.4, it's been in production use since March and iyt's very
stable:
- I have an agent on a queue, transferring a call received through
app_queue. the user presses #, hears transfer, types the
I don't think that is the case - if I add a wait(10) after the step 108,
i.e. the busy detection, the agent seems to be disconnected immediately at
the dial(), not after 10 seconds. That is what made me wonder what was
going on.
Yours
l.
On Sun, 01 Oct 2006 17:49:15 +0200, Adam Goryachev
at QueueMetrics. I have to patch asterisk _and_ use
MySQL? and JSP??? Good grief. Talk about overkill.
Doug.
-Original Message-
From: Lenz [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
I don't think there is. It would be rather overkill for what the app_queue
does; there are a number of queue stats packages, commercial and free,
that will provide a better approach to gathering stats for the purpouse of
running a call center or an inbound queue.
l.
On Fri, 21 Jul 2006
.
On Thu, 20 Jul 2006 16:56:06 +0200, Steven Totaro [EMAIL PROTECTED]
wrote:
I did a yum update queuemetrics and am now locked out of my box and my
unlimited license now shows as trial.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lenz
Hello list,
I am pleased to tell you that we have released a new version of
QueueMetrics. The main areas of improvement were the following ones:
- Bug fix: the Show members only button was not working in 1.2.0
- Improved graphical layout, gadget sizes and rendering in IE/Firefox
- Time zone
getting ours fully setup.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lenz
Sent: Wednesday, July 19, 2006 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] QueueMetrics 1.2.1
It really depends on the programming language you plan to use. I'd have a
look at the PHPAGI first, but there is not much as to AGI per se as with
the underlying programming language on one side and understanding Asterisk
on the other
Hope this helps
l.
On Mon, 10 Jul 2006
You can get the same result - well, maybe better - using a monitorin tool
that will display the availabe calls in the queue. QueueMetrics does it,
and FOP will do this too.
Hope this helps
l.
On Mon, 03 Jul 2006 15:09:44 +0200, Mailing Lists
[EMAIL PROTECTED] wrote:
Is it possible
I'm not sure you can do it if you want a play a file to the caller when
your agents pick up the line. I believe you may want to use a dial macro -
option M() - in the dial command that the local channel runs in order to
connect to the callback agent. see
after the end of the macro...
I only need to tell the caller that his line has been answered and that
he'll be able to talk...
Do I have to patch app_queue or is it already implemented somewhere,
maybe just a beep can be played to signal the bridge is done ?
Thanks,
Tristan
Lenz
Hello list,
I am pleased to tell you that we have released a new version of
QueueMetrics. The main areas of improvement were the following ones:
- Unattended client monitoring: for large call centers who work for third
parties, it is possible to have your clients log in directly to
Yes, it is definitely possible and a number of people use it this way, to
provide a kind of call-center virtual hosting. The functionality will be
expanded in the upcoming release, due by next week.
See http://queuemetrics.loway.it/manual.jsp the part about security keys.
Regards,
l.
In
I use an integrated DSL modem, print sharing, firewall, wifi and 2 SIP
port from DrayTek. Must be a version that has the firewalling without the
modem too. Quite cheap and worked very well for 2+ years.
l.
On Mon, 19 Jun 2006 21:37:39 +0200, Shaun [EMAIL PROTECTED]
wrote:
I'm looking
This should provide you enough information to get started.
http://www.astrecipes.net/index.php?q=astrecipes/compiling+asterisk+with+oh323
of course * can operate both SIP and h323 channels, but the support for
h323 (and I'd add, stability) is not the same you can expect with SIP or
IAX.
l.
Hello Christopher,
an Asterisk callback agent can be anywhere, even on a POTS number. He will
have to register with a number that can reach him as far as Asterisk is
concerned. I don't see the scenario you are proposing as particularly
difficult to implement in Asterisk.
Hope this helps
l.
Hello Leah,
it may be the quality of her link degrading - it happens easily with ADSL.
which error does she get? and she cannot receive calls at the same time,
right?
l.
In data Mon, 19 Jun 2006 22:52:40 +0200, Leah Newmark
[EMAIL PROTECTED] ha scritto:
We've been running an
You choose in the AgentCallBackLogin, while there is no need for it with
AgentLogin. A variable QUEUETRANSFER sets the context for # transfers.
Hope this helps,
l.
In data Tue, 16 May 2006 22:47:10 +0200, Thomas Kenyon
[EMAIL PROTECTED] ha scritto:
I know I must be being daft, but is
scritto:
I bet Signate will love this.
Lenz wrote:
Hello list,
we have prepared a short tutorial that will teach you to turn your
[EMAIL PROTECTED] box into a full-fledged call center within minutes, with
both always-on and callback agents available and the very extensive
reporting
Hello list,
we have prepared a short tutorial that will teach you to turn your
[EMAIL PROTECTED] box into a full-fledged call center within minutes, with
both always-on and callback agents available and the very extensive
reporting facilities that QueueMetrics provides.
You can download
Hello Kevin,
this is often a problem with MySQL disconnecting the client; we have an
updated version of the uploader script and we have released it a few days
ago - see http://queuemetrics.loway.it/download.jsp
I know the website is not very good at showing news, so we aplogize for
the
Hello list,
I am seeing that the current version of AMP / FreePBX sports app_icd
instead of app_queue as the default call distribution method - see
http://sourceforge.net/projects/icd
I was wondering if anyone had hands-on experience with app_icd and if
there are good selling points that make
Hello list,
we are testing an easier way to install QueueMetrics on an [EMAIL PROTECTED]
box (or any other CentOS/RHEL) using the yum package manager.
This is still experimental, so it may as well work as not work.
We are looking for testers who are willing to try this at home and any
Hello,
you can use the veryu same technique to turn recording on in any context -
what matters is that you bring along the uniqueid of the original call, so
you know how to match the different recordings you may find on your hard
disk!
l.
In data Wed, 12 Apr 2006 19:24:56 +0200,
That's a standard unix timestamp: # of seconds from Jan 1, 1970.
l.
In data Mon, 10 Apr 2006 08:55:49 +0200, [EMAIL PROTECTED] ha scritto:
Hi,
How do I read (make sense of) the timestamp in the queue_log? I'm
probably just slow but I don't understand it.
Thanks!
Regards,
Jan
--
Assum
On Thu, 06 Apr 2006 13:17:29 +0200, Tomislav Parčina [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
on a related note, we notice that if we've set atxfer = *1 in
features.conf
and blindxfer=#1, then attended transfers dont work. somehow, the Queue
app
I am not sure if it has been fixed, but using groups used to behave
erratically in earlier versions of *, so I am not used to doing it. A few
more configf lines are well worth the added stability.
l.
On Thu, 30 Mar 2006 10:30:50 +0200, Tomislav Parčina [EMAIL PROTECTED]
wrote:
In
You just add the same agent to both queues (don't use groups), like in
queues.conf:
[queue1]
member=Agent/101
[queue2]
...
member=Agent/101
Now Agent 101 is a member of both queues, and will not be called while
s/he is on conversation.
l.
On Wed, 29 Mar 2006 09:02:11 +0200,
of the queues... ie when
they are not here.
On 3/29/06, Lenz [EMAIL PROTECTED] wrote:
You just add the same agent to both queues (don't use groups), like in
queues.conf:
[queue1]
member=Agent/101
[queue2]
...
member=Agent/101
Now Agent 101 is a member of both queues, and will not be called while
s
Yes - I think so at least, this is surely true as of * 1.2.4, and have
seen nothing in the ChangeLog to suggest it has been changed.
By the way, this is usually an annoyance, as Agents would often like to do
an attended transfer, not an unattended one. Anybody has better solutions
to this
Just add the agent to both queues, * will take care of the rest.
l.
In data Tue, 28 Mar 2006 16:13:13 +0200, Matt [EMAIL PROTECTED] ha
scritto:
Hi,
What do I need to do to put an agent into two queues? The idea being
that the agent will get the call no matter which queue it comes into?
~
Hello Bjorn,
this will very likely be the solution to your problem:
http://www.oinko.net/astrecipes/index.php?n=119
Then you can use QueueMetrics - http://queuemetrics.loway.it - or
something similar to listen to both pieces of the call through the web
interface.
Hope this helps,
l.
In
In a inbound queue() environment, you can also have the same result using
QueueMetrics and any soft or hard phone.
l.
In data Tue, 28 Mar 2006 21:26:13 +0200, Bruno de Assumpção Loureiro
[EMAIL PROTECTED] ha scritto:
Does anyone know a softphone that can accept URLs during a call and
It is not so bad - it's the docs that aren't so clear - try this:
http://www.oinko.net/astrecipes/index.php?q=astrecipes/understanding+queue+logic
and beware of timeouts
l.
In data Fri, 24 Mar 2006 02:00:10 +0100, Douglas Garstang
[EMAIL PROTECTED] ha scritto:
Egads. Getting queues
Hello James,
you may be interested in knowing that QueueMetrics, even in the free
version, does exactly what you ask for, through a normal web browser
placed on each agent's machine. For details on the configuration, you may
want to contacty me off list. See the user manual at
Try setting it to sth like SIP/200 instead of a single number.
l.
On Mon, 20 Mar 2006 11:56:50 +0100, Christian B [EMAIL PROTECTED]
wrote:
Hello!
I'm trying to setup a perl-deadagi, but my perl skills lack. can
someone tell me why the following code doesn't work:
#!/usr/bin/perl
use
PROTECTED] ha scritto:
Lenz wrote:
I have added asterisks to denote a behaviour I dont understand; the
extension 101 is called twice in a row if 103 is unavailable. DO you
think this is a bug or there is a valid reason why * behaves like this?
(I'm running 1.2.4)
No, there is no bug here
Hello list,
after the last time that mpg123 wen ballistic on our production system, we
decided to skip mp3 playback altogether and to go for raw files. After
half an hour playing with mpg123 and sox parameters in order to translate
a mp3 file to a wav file that can be streamed back through
scritto:
Lenz,
This method is referred to as file-based or native MOH, and I have some
additional information regarding it. First, a short post on why we
moved from the rawplayer method to native MOH on our production box,
with a quote from Kevin Fleming regarding the impact the change
Hello list,
I have been researching a bit into the way the queue app works and how
different timeouts play together, and have prepared a short tutorial on
understanding queue timeouts - see
http://www.oinko.net/astrecipes/index.php?n=118 - any suggestion, error
found or correction is
Hi Arne,
what you write about seems to be mostly what Flash Operator Panel does.
Check it out before writing a clone yourself! :-)
l.
On Tue, 14 Feb 2006 13:21:37 +0100, Arne Morten Johansen [EMAIL PROTECTED]
wrote:
Hi there. We're going to develop a call centre app for internal use in
You'll have to use uattended transfers for CCs.
l.
In data Tue, 14 Feb 2006 09:37:24 +0100, Tomislav Parčina
[EMAIL PROTECTED] ha scritto:
When agent tries to transfer a phone call (*2 - att transfer) he hangs
up. Why? When a phone call isn't from queue then att transfer works fine.
In
I believe many users would benefit from such an improvement. That's the
kind of annoyance that is usually not present in a decent PBX and that
people have the worst time getting used to (at least in my experience).
l.
On Mon, 13 Feb 2006 14:46:35 +0100, Alex Barnes
[EMAIL PROTECTED] wrote:
Hello list,
as likely most of you, this morning I have compiled my fresh 1.2.4 with
H.323 support.
I have prepared a recipe of the compilation process so that maybe it can
be useful for future reference or for a quick start:
http://www.oinko.net/astrecipes/index.php?n=102
Any comment or
On Sat, 31 Dec 2005 20:23:09 +0100, BJ Weschke [EMAIL PROTECTED] wrote:
From the Asterisk CLI you can do show queues and show agents.
There are also a number of third party tools, free and not-free, to
take information from Asterisk and present it in real-time and on a
historical basis.
http://www.gizmoproject.com/
from the website, it quite looks like skype - no network setup, IM
integration, you may call POTS phones by paying the company who did it.
not very useful, in the end, if the purpouse is asterisk-skype
interoperability - I doubt that every one of the millions
You may want to check this out:
http://www.digium.com/asterisk_handbook/agentlogin_queues.html
A number of our clients use it to analyze outgoing calls from within
QueueMetrics.
Yours
l.
In data Mon, 19 Dec 2005 20:57:54 +0100, Jim Miller
[EMAIL PROTECTED] ha scritto:
Is there a
Hello list,
I am pleased to tell you that we have released a new version of
QueueMetrics. This time QueueMetrics comes of age, as we have almost
reached version 1.0. :-)
This version is the result of a feature-freeze of version 0.9.7, and
vastly improves memory efficiency, letting you
Yes, it is correct. The best way to handle this problem (on 1.2) is to
pause the agent before the outbound call and the unpause him when he's
done.
Yours
l.
On Tue, 13 Dec 2005 15:20:56 +0100, Patrick Fortin [EMAIL PROTECTED]
wrote:
Hi
We have a call queue setup with several agents
Hi Rob,
you could build a simple Perl or Python script to create incoming calls
using callfiles. We have used such a strategy and it seems to be working.
l.
On Thu, 08 Dec 2005 14:15:50 +0100, Rob Hillis [EMAIL PROTECTED]
wrote:
I'm currently starting development of an add-on to a program
Hi,
I don't think it is impossible, though not yet supported by Asterisk
out-of-the-box. You could have a general queue plus a queue per each
agent, and you would route the call to each agent based on the caller*id.
This might end up spoiling the advantage of a queue, meaning that you
there should be a way in agents.conf to autologoff agents after a while
the do not answer the phone.
l.
In data Sat, 03 Dec 2005 23:48:05 +0100, Vladimir S. Blazhkun
[EMAIL PROTECTED] ha scritto:
-- Called 1101
-- Agent/1101 is ringing
-- Got SIP response 480 Temporarily Unavailable
You'll have to have agents join the queues by issuing the commands
AgentLogin() or AgentCallBackLogin() from an extension in your dialplan.
l.
On Thu, 01 Dec 2005 17:53:19 +0100, gc [EMAIL PROTECTED] wrote:
Thanks. I made change to joinempty=yes. And now I can hear the music on
hold.
One simple way to overcome this problem would do to make an attended
transfer to check whether the receiving person is available and willing to
take the call, and then an unattended transfer to discharge the operator
of the call.
l.
On Fri, 02 Dec 2005 16:21:39 +0100, James Armstrong
Hi Trey,
It is done automagically by the system - see the setting named announce
in the queue definition.
Hope this helps
l.
On Wed, 30 Nov 2005 20:38:20 +0100, Trey Blancher
[EMAIL PROTECTED] wrote:
I want to play a file for an agent that answers a queue call, before
the agent is
hello list,
inspired by a number of previous posts, I prepared this node in the
AstRecipes wiki:
http://www.oinko.net/astrecipes/index.php?n=107
It wants to be a good starting point to show low-level hardware problems
with zaptel (and PCI in general) cards.
Anybody has suggestions to be
Hi Johann,
we engineered QueueMetrics out of the queues of * version 0.7, but never
found that origposition argument. And it's not present in our current 1.2.
Where did you find it?
Yours
l.
In data Tue, 29 Nov 2005 19:57:47 +0100, Johann
[EMAIL PROTECTED] ha scritto:
Entries in the
Hello list,
this is just an announce of a new mailing list dedicated to deploying,
running and managing real-
world Asterisk-based call centers. The mailing list is in English and
allows knowledge sharing
for this very important - and yet somehow less considered - Asterisk
deployment area.
PROTECTED]
wrote:
I've asked the same question in several occasions in the past and never
received a response. I figured this project was dead and stop pursuing
using it.
- Waldo
On Nov 21, 2005, at 12:17 PM, Lenz wrote:
Well, this is interesting - is anybody actually using app_icd out
Hello Jason,
if the system is so simple, why don't you connect the queue straight to a
couple of you terminals, i.e. not to Agent/101 but to SIP/214. This way
you have no login/logout.
Yours,
l.
On Tue, 22 Nov 2005 16:20:50 +0100, Jason Lixfeld
[EMAIL PROTECTED] wrote:
Here's what
AFAIK there were some known issues preventing call transfer from H323
terminals, at least with Innovaphone ones.
Yours
l.
On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao
[EMAIL PROTECTED] wrote:
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls
can be
Well, this is interesting - is anybody actually using app_icd out there?
:-)
l.
On Thu, 17 Nov 2005 00:54:56 +0100, Tyler [EMAIL PROTECTED] wrote:
Anyone using app_icd? I need to use some of the advanced features that
the regular asterisk Queue() application won't provide. Anyone have
You could use a prefix-based agent numbering scheme, like Agent/XXYYY
where XX is your customer code and YYY their own agent number. When
showing activity to a customer, you strip the XX part or you may leave it
alone, as it makes no big confusion to the client.
Yours,
l.
On Fri, 18
rm -rf for the following dirs (taken from
http://www.oinko.net/astrecipes/index.php?n=93 ):
/etc/asterisk
/var/log/asterisk
/var/lib/asterisk
/var/spool/asterisk
/usr/lib/asterisk
any other?
l.
On Tue, 15 Nov 2005 11:02:08 +0100, Matteo Piazza
[EMAIL PROTECTED] wrote:
Is there a
Hello James,
you could approach this problem in many a way. I'd suggest to make your
support guy log on to the queue using AgentCallBack and enforce
joinempty=no in the queue itself. When your agent goes to lunch, he logs
off and people cannot join the support queue anymore, so you move
Hello List,
I'm glad to announce that we have released the first version of
QueueMetrics that supports MySQL storage of queue_log data. It is still
experimental, so if you run such a setup and would like to give it a try,
you are welcome. The MySQL adapter should adapt to any existing table
Hello list,
we are in the process of releasing a new version of QueueMetrics that will
be able to analyze queue_log data stored on MySQL table, with no need to
change your table definition. If you currently host queue_log data on
MySQL and would like to help us testing it, please drop us a
Hello,
did you try using a Local/XXX channel? it should work!
l.
On Tue, 01 Nov 2005 15:10:03 +0100, Stefan Günther
[EMAIL PROTECTED] wrote:
Hi,
I want to transfer a call that has come into one queue, and that I have
already accepted, into another queue.
When I try this asterisk tells
If you can, avoid it: you want to report what *people* are doing, not
telephone terminals. You lose a lot of flexibility using telephones. Use
PersistentAgents instead!
Bye
l.
In data Fri, 28 Oct 2005 14:51:56 +0200, Arne Morten Johansen
[EMAIL PROTECTED] ha scritto:
What about
Hi Alessio,
The opposite of logging in with Agentlogin is simply hanging up the phone!
:-)
If you use AgentCallBack, you can instead logoff explicitly.
You vcan also log off users manually from the console.
Hope this helps
l.
On Tue, 25 Oct 2005 12:00:07 +0200, Alessio Focardi
[EMAIL
Hello,
you should look at voicemail.conf, see emailsubject and emailbody. I
believe that it can handle Chinese as any other language as well, as you
can specify the charset encoding.
Bye
l.
On Mon, 24 Oct 2005 10:03:22 +0200, Ronald Wiplinger [EMAIL PROTECTED]
wrote:
I was looking for
Hello,
I usually use
exten =
s,1,SetVar(MONITOR_FILENAME=/var/spool/asterisk/q/QSAMPLE-${UNIQUEID})
exten = s,2,Queue(q-sample|nt|||60)
and it seems to work, then use QueueMetrics to keep track of who was
talking to whom, instead of using the Agents monitoring.
Bye
l.
On Sat, 22 Oct
This should give you a guide.
http://www.oinko.net/astrecipes/index.php?n=102
I have been using both H323 and OH323 with no big problems since * 0.7.
The only thing you notice is an added need for restarting * on busy
machines.
Bye
l.
On Thu, 20 Oct 2005 14:23:53 +0200, Carlos Arnt [EMAIL
Hello Waldo,
if you use AddQueueMember plus a fake queue_log registration, you can tell
who the agent was, not just from what terminal she was connecting from. It
is then possible to report who was available at a certain time, or see
agents logging on and off, going to pause, measuring the
with
it, but this is not likely the best way to go. :-)
Bye
l.
On Fri, 21 Oct 2005 10:23:55 +0200, Waldo Rubinstein [EMAIL PROTECTED]
wrote:
Lenz,
Thanks for the response. I agree with you. However, I have a couple of
questions:
1) How to do a fake queue_log registration
2) One of the needs I have
Hi Matteo,
it looks really promising. I'll give it a try!
l.
On Wed, 19 Oct 2005 23:38:00 +0200, Matteo Brancaleoni
[EMAIL PROTECTED] wrote:
Hi to all,
sorry for crossposting the -dev and -user lists, but I think this could
be quite interesting news for EuroISDN people, expecially BRI
Hello Jeremy,
I have been using the MP-108's with H323 interface in a project over one
year ago and I found them to be quite good and easily interoperable. After
a while both units seemed to lose the IP address when turned off, while
retaining other parameters, so it's quite a nuisanmce,
Hello,
if you use a mechanism like agents, * will know that there is nobody at
the first level of penalty and route the call to the other level. A
different approach could be to have a queue ring A for say 20 second,
timeout, route the call to a second queue where B and C are. This should
Hello,
you should use asterisk agents and you'll see that the problem will go
away.
Bye
l.
On Tue, 18 Oct 2005 14:13:32 +0200, [EMAIL PROTECTED] wrote:
Hi,
I'm running 1.2 beta1 in a mini call center.
I have 3 queues with 10 operators, and I'm running into some trouble
because when
Hello,
are you using Asteriks agents or dialing straight to extensions? because
if you are using agents for incoming calls and then you dial straight
out of Asterisk, Asterisk will not know that the agent is busy. One
possible workaround would be to make a call to the agent using a .call
.
Hence the proper behavior has to come through feature request only.
-- jt
On Mon, 2005-10-17 at 04:30, Lenz wrote:
Hello,
are you using Asteriks agents or dialing straight to extensions? because
if you are using agents for incoming calls and then you dial straight
out of Asterisk, Asterisk
Hello list,
I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with
a TDM400 card and H.323.
You can find it at http://www.oinko.net/astrecipes/index.php?n=102
Any comment / suggestion / modification /bugfix is welcome!
I was wondering: is there any way to build a version
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