In data Mon, 10 Oct 2005 23:54:43 +0200, tim panton
[EMAIL PROTECTED] ha scritto:
Errm, I had not intended that URL to be made public to all of the 1
subscribers to
this list! My voicemail will be overflowing :-( So I've removed it for
now.
I will notify the list when it is ready
Hello list,
I am looking for a way to have multiple remote Windows users download a
package and get connected to *. My idea would be that they run a simple
app, it connects without any setting to an * box (maybe via IAX) and then
people press a button to talk. It would be okay if they had
Hello list,
today I have been busy playing with addQueueMember, and it is well known
that it does not log to the queue_log file.
The answer - bad as it may seem - is to add a fake queue_log data for each
logon and logoff. This was covered previously by
In data Mon, 10 Oct 2005 18:57:20 +0200, Michael Van Donselaar
[EMAIL PROTECTED] ha scritto:
Rather than recompile with presets, you'd probably want to change the
reg keys
used in the installer. When I was first developing iaxComm for family
and
friends, I distributed the executable with
In data Mon, 10 Oct 2005 14:03:55 +0200, tim panton
[EMAIL PROTECTED] ha scritto:
Yep, I'm working on such a thing.
I have a demo version running at http://www.westhawk.co.uk/software/
faceless/CallUs.html
You don't even need to install it, it runs in the user's browser.
( you will need IE6
QueueMetrics does that, together with a bunch of other things (like
showing pauses, logons, calls waiting and more) - see
http://queuemetrics.loway.it
Or you can use the managemente interface and poll it every once in a while.
Bye
l.
In data Wed, 28 Sep 2005 13:02:31 +0200, Mark Elkins
You should either use Agents (standard or callback) or disable voicemail
on the second server, with a straight dial instead of the dial+voicemail
macro you'll likely be using.
bye
l.
In data Fri, 23 Sep 2005 17:15:38 +0200, [EMAIL PROTECTED] ha scritto:
I all.
I have configured a pair
Hello Matt,
very interesting setup! are you using asteriak queues for inbound or not
at all?
Bye
l.
In data Thu, 22 Sep 2005 06:25:44 +0200, Matt Florell [EMAIL PROTECTED]
ha scritto:
We wrote VICIDIAL(part of the GPL astGUIclient suite
http://astguiclient.sf.net) for our call center
Hello,
is there a best practice to upload queue_log file into MySQL? or - better
- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
Thanks
l.
--
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by
AM, lenz wrote:
Hello,
is there a best practice to upload queue_log file into MySQL? or -
better
- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
Thanks
l.
--Assum est, versa et manduca.
___
--Bandwidth and Colocation
Hi,
QueueMetrics version 0.9.5 rc 2, out today, does the trick and allows
agent pause monitoring (together with the rest of the stuff).
See http://queuemetrics.loway.it
Thanks
l.
In data Wed, 14 Sep 2005 07:28:51 +0200, Callum McGillivray
[EMAIL PROTECTED] ha scritto:
Hey Kevin,
Hello list,
while using the new Asterisk 1.2 beta, I keep noticing this when an agent
transfers a call from a queue to another extension:
[Except from queue_log]
1125912636|1125912630.134|queue-dps|NONE|ENTERQUEUE||21
1125912638|1125912630.134|queue-dps|Agent/101|CONNECT|2
Hello,
after a while that it has been forcibly down, the ASTERISK-ITA mailing
list, dedicated to general Asterisk discussion in Italian, is now back
online.
The list is located at http://groups.yahoo.com/group/asterisk-ita
Thanks for your interest
l.
--
Assum est, versa et manduca.
Hello list,
I put together a quick note about how to see oh323 calls while they are
handled by your * box.
http://www.oinko.net/astrecipes/index.php?n=89
The article is just a draft with usage examples; I'd love to hear your
comments and updates if there is something I got wrong.
Thanks
channels
Thanks
Giles
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: 31 May 2005 12:51
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] monitoring oh323 calls
Hello list,
I put together a quick note about how to see oh323 calls
while
Hello list,
I am glad to announce that XC-AST version 0.9.0 is out today.
New functionalities include:
* Though not yet available to the end user, this release inclued the basis
of the Outbounds Call Manager that will be released for 1.0. If you update
from a previous version, have a look at
Hello list,
I have spent the last couple of days installing a network of Linksys
PAP2-NAs and some Welltech LP302s linked to an * box. I have found that
they work great and have posted the configuration files on AstRecipes, so
they can be shared.
See
Hello list,
is there a way to configure a SIP phone to autodiscover its own PBX ona
LAN? when running H.323, it is quite trivial to set up the Gatekeeper
autodiscovery so that you can have all units working with dynamic DHCP
addresses and you do not have to configure each unit by hand when
Hello,
from what I see, I guess they're only ways to insert a piece of speech
without recording it; you could easily record the phrases yourself and add
Playback()s instead.
BTW, I'd like to thank Tim for sharing his recipe with us. Anybody else's
got a recipe to share? :-)
l.
In data Sat,
In data Tue, 26 Apr 2005 09:13:52 +0300, Tzafrir Cohen
[EMAIL PROTECTED] ha scritto:
One clarification:
On Mon, Apr 25, 2005 at 05:07:31PM +0200, lenz wrote:
See http://www.oinko.net/astrecipes
All content is licenced as creative commons, so if you got a recipe to
spere, feel free to post
Hello,
if anyone is interested, there is a new wiki about Asterisk recipes,
i.e. step-by-step descriptions on how to perform something with your *
box. This is quite different from most * sites around, that are either
questions-and-answers forums or are dedicated to documenting a feature.
In data Mon, 25 Apr 2005 17:33:14 +0200, Bruno Hertz [EMAIL PROTECTED] ha
scritto:
Good idea, but don't we have already the Wiki tips/hints, editable by
anybody ? I understand people like to contribute, which is great. But
spreading the info all over the web instead of centralizing it might
be
In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield
[EMAIL PROTECTED] ha scritto:
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED]
wrote:
Hello,
if anyone is interested, there is a new wiki about Asterisk recipes,
i.e. step-by-step descriptions on how to perform something
/25/05, lenz [EMAIL PROTECTED] wrote:
In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield
[EMAIL PROTECTED] ha scritto:
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED]
wrote:
Hello,
if anyone is interested, there is a new wiki about Asterisk
recipes,
i.e. step-by-step
Hello list,
I am glad to announce that it is now possible to try XC-AST, the queue_log
file analyzer implementing most call centre metrics for the app_queue,
using a demo password.
See http://demo.xcept.it/xc-ast/xcast-live.jsp
Some people complained that it was quite too complex to set up a
not seem to be working, I am doing something wrong. It is
constantly complaints that file placed in 'File' field is not found.
Please
let me know how to resolve this.
Thanks
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: Thursday, April 14
In data Thu, 14 Apr 2005 10:39:38 -0700, Richard Lyman [EMAIL PROTECTED]
ha scritto:
lenz wrote:
Hello,
you have to enter /var/log-xcast/queue_log_live as the file and DPS
as the queue (select it from the drop-down box) for the demo to find
*snipped
one thing i noticed, i did a copy paste
In data Thu, 14 Apr 2005 10:39:38 -0700, Richard Lyman [EMAIL PROTECTED]
ha scritto:
lenz wrote:
Hello,
you have to enter /var/log-xcast/queue_log_live as the file and DPS
as the queue (select it from the drop-down box) for the demo to find
*snipped
one thing i noticed, i did a copy paste
Hello,
I don't think this is an * issue, it was more likely the telecom you
connect to having some kind of issues. If they hacked your * box, they
would not likely call you and tell you. Unless they have some reason to
show you down and they made the call themselves.
l.
In data Thu, 14 Apr
Hello,
most phone models have a way of setting the maximum incoming call limit,
so I guess that if you set it to 1 the phone will signal busy when a
user is talking.
Another alternative would be to set up your queue using Agents; in this
case * knows whether an agent is busy or not and will
Sure: add the endpoint directly in queues.conf instead of putting
Agent/xxx.
Bye
l.
In data Fri, 1 Apr 2005 12:57:32 +0200, Obihuan [EMAIL PROTECTED] ha
scritto:
Hello all,
There are any way for the queue agents in asterisk that they do not
need to login in the queue to begin recibing calls?
Hello,
XC-AST, even in the free version, does the trick no matter what telephone
terminal you are using. Give it a try.
BYe,
l.
In data Sat, 19 Mar 2005 20:30:48 +0100, Vikram Rangnekar
[EMAIL PROTECTED] ha scritto:
+++ James Coberly [18/03/05 10:28 -0700]:
Try DIAX.
In data Fri, 01 Apr 2005 11:09:24 +0800, El Flynn
[EMAIL PROTECTED] ha scritto:
Matt Roth wrote:
Preferably, I would like an out-of-the-box solution, but custom-coding
is an option as long as the necessary data is available from Asterisk.
If anyone could point me in the right direction, it
Hello,
I have set up Comedian Mail on my Asterisk system.
I am using Voicemail not Voicemail2 in my extensions.conf file.
The system works great except for 1 thing...It is not possible to create
custom unavailable or greeting messages for 3/4 voicemail boxes.
For some odd reason 3/4 users are
Anybody got a few decent pointers to get people started with ICD? We'd
like to integrate that in Xc-Ast too.
Cheers,
l.
In data Tue, 15 Mar 2005 18:12:18 +, Asterisk [EMAIL PROTECTED] ha
scritto:
Is there any development ongoing with ICD ? I wouldn't want to get
involved in something
In data Tue, 15 Mar 2005 17:45:18 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] ha scritto:
Any real experiences with * on this?
You can create a quite flexible callcenter solution from ICD (search for
app_icd). It is more of a framework to create a call center solution than
a finished product.
is this what you are looking for?
exten = _0XX,1,Dial(OH323/[EMAIL PROTECTED])
Bye
l.
In data Fri, 11 Mar 2005 11:08:59 +0200, Altus Snyman
[EMAIL PROTECTED] ha scritto:
Goo day all
This is our setup
Client phone--(SIP)--asterisk server---SIP/IAX---asterisk---
-- goes out to
Hello,
you just need something very simple for the agents to be registered, like
[ed-agenti]
exten=s,0,Answer
exten=s,1,AgentLogIn()
You should then configure a queue and agents.conf.
A sample (and simple) queue configuration is described at
Hello list,
I am glad to announce that XC-AST version 0.8.0 is out today.
This version introduces an importante new feature: you can now listen -
using your browser - to calls that have been previously monitored. This
way you can click on a call and listen to the actual conversation, and see
XC-AST does that too, no matter what the phone used is. It lets you see a
lot more of calls being handled and produces nice reports.
See http://demo.xcept.it/xc-ast
Bye
l.
In data Wed, 2 Mar 2005 17:18:24 -0500, mattf [EMAIL PROTECTED] ha
scritto:
We use astGUIclient suite, it has this
] On Behalf Of lenz
Sent: Jueves, 03 de Marzo de 2005 10:43 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk URL and Callcenter Apps
XC-AST does that too, no matter what the phone used is. It lets you see a
lot more of calls being handled and produces
Hello,
WavePad worked perfectly in the free version. Thank you.
l.
In data Sat, 26 Feb 2005 11:47:56 -0500, mattf [EMAIL PROTECTED]
ha scritto:
The free utility WavePad for Win32 will play and edit GSM files as well:
http://www.nch.com.au/wavepad/
To convert to/from GSM on Win32 you can use
Hello list,
I am having trouble listening to GSM files created by Asterisk using a
browser. I am noticing that some of my users succeed in listening to them
and some others don't. I guess it is something of a codec problem that
does not seem to be installed on all machines (though they are
Hello John,
xc-ast - http://demo.xcept.it/xc-ast - is currently able to produce
reports based on the file-based queue_log but will soon be able to run the
analysis on a mysql database (this feature is planned for 0.9). We will
provision a perl upload utility, though it will not initially be
Hello list,
the guys here at Xcept wanted to say Merry Christmas to the list and the
people who make Asterisk possible, so we put together this little
Asterisk-based Christmas app...
Justr dial IAXtel 1-700-444-6295 and be happy :-)
Yours,
lenz, laura and marco
the crew at Xcept
--
Creato
Yes, of course you can do that. I have this very setup working for the
office, with * aggregating voip and isdn incoming calls and forwarding
them to my laptop wherever I am.
just follow the instructions on the FWD website, and run iax2 debug from
the console to see what's happening in
Hello list,
I am looking for some ActiveX or - better - Java client that I can embed
in a web page and connect to Asterisk. I would like to implement a click
here to talk to a live operator button to be put on a web page. Is there
something already available, both in the commercial and free
If anybody is interested, we are starting an asterisk mutual support
mailing list in Italian.
The list is located at
http://www.smartgroups.it/mailman/listinfo/asterisk-ita and managed with
GNU MailMan, so most users should be already familiar with it.
Thanks
l.
--
Creato con M2, il
Hello,
we make XC-AST and can install it for you, or we can help you installing
it. How big is your call center? Under which environment did you try to
install it?
Thanks
l.
In data Fri, 10 Dec 2004 14:58:09 -0500, John Bittner [EMAIL PROTECTED] ha
scritto:
I have spent the last 3 days
try sj phone, it works fine for me. works on win32 and mac, not sure if
there is a linux port too.
http://www.sjlabs.com/
l.
In data Wed, 01 Dec 2004 11:42:14 +0100, Tomasz Chmielewski
[EMAIL PROTECTED] ha scritto:
Hello,
Is there a list of software phones which will work with Asterisk?
For
I usually program queue_app so that a queue times out after a few minutes.
After that I send a message telling people to press 1 to leave a voice
message and 2 to wait again; if they press 1 they get voicemail, if they
press 2 they re-enter the queue system.
it should be quite
In data Mon, 29 Nov 2004 08:51:08 +, Jean-Michel Hiver
[EMAIL PROTECTED] ha scritto:
It's a very interesting idea. The more I think about it, the more I
wonder . . .
Sounds like something that would end up being used as a dating phone
service to me :-)
yes, I think so. that could be a
Hello list,
I am glad to announce that XC-AST 0.5, released today, offers real time
queue monitoring facilities that let you see the calls flowing through a
set of Asterisk queue(s) and agents logging on and off.
This way, XC-AST provides an one-stop solution to generate reports,
monitor
hello list,
I was wondering: anybody ever wrote an asterisk based bbs? not a bbs about
asterisk, but a vocal bbs that runs on asterisk, so that people can call,
hear the discussion of the day, leave messages, etc.
it seems a rather basic application to me though I cannot find much about.
In data Sun, 28 Nov 2004 16:55:06 +0100, Michael Vogel [EMAIL PROTECTED] ha
scritto:
lenz schrieb:
I was wondering: anybody ever wrote an asterisk based bbs? not a bbs
about asterisk, but a vocal bbs that runs on asterisk, so that people
can call, hear the discussion of the day, leave
Hello,
I sometimes find AGENTDUMP rows on the queue_log; anybody knows what do
they mean?
Thanks
l.
--
Creato con M2, il rivoluzionario client e-mail di Opera:
http://www.opera.com/m2/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
In data Fri, 26 Nov 2004 23:17:27 +0800, el Flynn
[EMAIL PROTECTED] ha scritto:
lenz wrote:
Hello,
I sometimes find AGENTDUMP rows on the queue_log; anybody knows what
do they mean?
Thanks
l.
AGENTDUMP: The agent dumped the caller while listening to the queue
announcement.
You can find
Hello,
I have come across some weird behaviour in the queue_log file when using
AgentCallBackLogin, things like:
1100214870|1100214864.141|NONE|Agent/101|AGENTCALLBACKLOGIN|[EMAIL PROTECTED]
...
1100215527|1100215501.194|NONE|Agent/101|AGENTCALLBACKLOGOFF|[EMAIL
PROTECTED]|657|Autologoff
...
Hello,
You can of course play whatever message you want before moving the caller
to a voicemail directly from the dialplan, as follows
exten = s,3,Playback(voicemail-invitation)
exten = s,4,VoiceMail,s2001
This way you can setup custom messages.
Hope it helps
l.
In data Mon, 1 Nov 2004
I know Innovaphone - http://www.innovaphone.de - has nice (but not cheap)
H.323 DECT gateways that support multi-cell roaming.
l.
In data Mon, 01 Nov 2004 11:20:06 -0800, TC [EMAIL PROTECTED] ha scritto:
-Original Message-
I am able to buy and test a wireless basestation that can
better.
Jeremy
On Fri, 29 Oct 2004 21:23:54 +0200, lenz [EMAIL PROTECTED] wrote:
Hello list,
I'd like you to know that version 0.3.5 of XC-AST is out - now it is all
translated into English and has a 20 page user manual, so I guess it's a
bit more user friendly. See http://demo.xcept.it/xc-ast
Hello list,
I'd like you to know that version 0.3.5 of XC-AST is out - now it is all
translated into English and has a 20 page user manual, so I guess it's a
bit more user friendly. See http://demo.xcept.it/xc-ast
Plans for the future include a real time queue monitoring feature; I was
Hello list,
I was looking for a way to implement non-blind call transfers with *, i.e.
the usual behaviour of most PBXs when pressing the flash button:
- A and B are talking
- A pushes flash
- A is free to compose a new number
- B hears music on hold
- A's call is answered by C
- A hangs up
- B
maybe you should create a queue and add the people whose phone should ring
to the queue.
we have a number of such small queues at the office and they work just
fine.
hope this helps
l.
In data Tue, 12 Oct 2004 11:20:22 +0200, Altus Syman
[EMAIL PROTECTED] ha scritto:
Good day all
We have
Hello list,
I'd like to use the Send URL function in the Queue command, so that when
an agent answers a call s/he sees the browser opening with a page having
pertinent information. Anybody can tell me of a soft phone supporting this
feature under Windows? I tried with SJPhone with no
Hello list,
I have upgraded the queue_log analyzer I was talking about last month so
that some sample pages are available in English too, and this time I
tested them with Firefox too. :-)
This is mostly an incremental upgrade meant to show what the program is
currently doing. I expect to
In data Wed, 18 Aug 2004 16:18:48 -0400, Steve Szmidt [EMAIL PROTECTED]
ha scritto:
Thanks. It works fine in Opera and IE but I guess I'll check it with some
Mozillas.
BTW the windows server is a high-reliability cluster of Linux boxes
running Apache Tomcat I have parked the page on. :-)
Thu, 19 Aug 2004 09:42:04 +0200, lenz [EMAIL PROTECTED] ha
scritto:
In data Wed, 18 Aug 2004 16:18:48 -0400, Steve Szmidt [EMAIL PROTECTED]
ha scritto:
Thanks. It works fine in Opera and IE but I guess I'll check it with
some Mozillas.
BTW the windows server is a high-reliability cluster
Hello,
the analyzer is written as a Java model-2 webapp, that runs on any Java
application server, like Tomcat or WebSphere, using an in-house
developement architecture and MySQL for user authentication. I usually
develop it on Windows and deploy on Linux, but you can choose freely. What
I
Thu, 29 Jul 2004 15:59:39 +0200, lenz [EMAIL PROTECTED] ha
scritto:
Hello list,
as I'm writing a little perl parser for queue_log analysis
--
Creato con M2, il rivoluzionario client e-mail di Opera:
http://www.opera.com/m2/
___
Asterisk-Users mailing
Hello,
is there a way I can obtain the IP endpoint address when the telephone is
called from app_queue?
I even tried creating a pseudo number, so that instead of having my queue
call straight (say) OH323/1234 I call a number on asterisk where I log the
call id and then do the dialling. Of
Hello list,
as I'm writing a little perl parser for queue_log analysis, I'd like to
know *which* telephone answered a specific queue call. Unfortunately
app_queue only logs the call id but does not log the call end point. This
is okay for SIP endpoints, because their call id is something
Hello all,
I need to write a queue_log parser that is going to implement more or less
the functionalities described here
http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html
of course not everything from scratch, but this is where I'd like it to go.
I am looking for
-
I have been using a Prestige 2000W, but I'm not really satisfied with it.
The software is buggy and the telephone menus are very slow; sound quality
is ok if you're in the same room as the access point, but degrades quickly
if you go outside. Batteries don't last much and when I leave the
as an alternative, you can use dial with the T or t option and then have
the user press # for a blind transfer; this way you can manipulate
extensions easily.
l.
ps. would surely be nice to have Dial() be able to recognize other
extensions, just like it now doe4s with #.
l.
In data Wed, 07
Hello list,
I wonder if this is possible with Asterisk:
- While talking through Asterisk, I would like a client to start
recording a call by typing, say, #99#
I know it is possible to do it using an external monitoring application,
but I want to know if it's possible to have Asterisk
(I posted this note on
http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too)
I tried to put together comments that were asked for on the P2000W.
These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott
for getting me started with it.
DTMF keys work fine and are read
I could not find the firmware for the 2000W on ZyXEL's websites (I tried
the italian one, the swedish one, the english one and likely some other
too - the 2000w seems to be completely absent from their support section),
but it is available here:
Hello,
I am having trouble registering a SIP telephone Zyxel Prestige 2000W to my
Asterisk PBX. I am rather confident that Asterisk is working as there are
other H323 telephones working fine and a couple of SJPhones working
correctly.
I keep getting:
Jun 24 10:31:08 NOTICE[-1254138960]:
setting
Default Voice codec G.729 8k
DTMF relay outbound
Hope this helps
Cheers
Giles
- Original Message -
From: lenz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 24, 2004 10:10 AM
Subject: [Asterisk-Users] Cannot register Zyxel Prestige 2000W
Hello,
I am having trouble
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