).
Clarification: Asternic Call Center Stats Lite is free (GPL3) and can be
downloaded from the above link. The PRO version is commercial.
Asternic CDR reports for FreePBX is also free and available for download on
the asternic.biz site.
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I'm speaking at ElastixWorld: http
On Mon, Oct 5, 2009 at 6:31 PM, Danny Nicholas da...@debsinc.com wrote:
$595 US. Cheap, but depends on the price of local dirt.
LOL... dirt in Argentina is cheaper.
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Hello,
Just a correction, Asternic Call Center Stats is not from
asteriskguru. Asteriskguru has its own statistic program that is not
open source, but free to use. Asternic was written by me (not
asteriskguru) and has an open source version and a commercial one.
Best regards,
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As for queue_log analyzers, you can also look at
http://stats.asternic.org/ . I do not want to give you an opinion
because I wrote it myself. There is a fully functional free version.
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On Mon, May 4, 2009 at 9:15 AM, Louis-David
on and a very confused customer.
I certainly would not rule out Asterisk, but you always start with the
cables.
I have just seen this myself on a pure voip and busy callcenter...
with recordings to prove it too. I would not rule Asterisk either, as
I do not have many cables to check!
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Asterisk environment.
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Then you will want to move the file to /etc/rc.d/init.d
And add it to startup with:
chkconfig --add op_panel_redhat.sh
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Hello,
I have finally released the queue stats package to the public.. please go to:
http://www.asternic.org/stats
To get it or see the online demo.
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to perform transfers,
dragging the other leg (default behaviour), or your own.. you have to
select which one you like with reverse_transfer=1 in op_server.cfg
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Context=my-agent-context
And it will *probably* work... I say probably because local channels
are a #!@ to track with manager events. Look at the source, there is a
function named local_channels_are_driving_me_mad
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native attendant transfers with asterisk 1.6.
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Hello,
In short – Asterisk is not able to recognize that the 'other' person to whom
call was made has hung up – hence the channel stays busy.
http://kb.digium.com/entry/1/6/
I would try with busydetect and busycount.. Best regards,
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a scroll on this to display everything? do i need to
resize the buttons?
For sure someone now how to solve this basic question:)
You can reduce the button size, update to the latest snapshot that
includes 'slow' horizontal scrolling, or both. Best regards,
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Hudlite can do this we caller ID, but the DID feature is not
there yet.
Does anyone have any other software they know of that can do this?
FOP can do that. http://www.asternic.org
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can join the Meetme room muted...
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to improve the documentation if you feel is
innacurate or incomplete, make translations, etc.
And if you do not have time to do custom mods, you can hire someone to
do it for you... ;)
Best regards,
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they've stopped working.
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on this interfaces should be usability and flexibility
and not only looks.
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listening to the events.
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' function but is marked as post 1.4 (
http://bugs.digium.com/view.php?id=5841 ). This is a very much needed
feature.
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and
displays status of your asterisk queues. There are also soft phones
that can handle url passing, some feature complete callcenter packages
(astguiclient). You can also do it yourself using the manager API or
custom AGI scripts. Regards,
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On 7/30/06, Dovid Bender [EMAIL PROTECTED] wrote:
any programs out there that do this ?
Dovid
You can use FOP: http://www.asternic.org
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asterisk
you have to drag the phone icon to the
destination. Anyways, please check FOP archives or subscribe to the
mailing list as this is related to FOP and not Asterisk itself.
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then.
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routines. The
most recent astcc found in svn includes the code to ignore the HUP at
the top of the script. Be sure to use that version. Regards,
On 6/25/06, Nicolás Gudiño [EMAIL PROTECTED] wrote:
Hi Ronald,
If a user calls and hangs up before the destination party rings, than
the in-use flag
a call. The suggesion was to
catch the SIGHUP signal in the process and ignore it. I wrote the following
piece of code at the star of the agi.
You might want to read this:
http://bugs.digium.com/view.php?id=6491
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://www.asternic.org
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to differentiate them in an easier way than fully listing.
The only reserved characters for regexp buttons are underscore and
ampersand. Any other character should be treated as per PERL regular
expressions. Anyways, your milleage might vary, you need to try it
out.
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are crafted.
Read the documentation, not the readme, at http://www.asternic.org to
get an idea of the button types and how to configure them. There is
also a low traffic FOP mailing list you can subscribe from the same
page.
Regards,
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, profiling and optimizations.
The upgrade instructions are on the tarball UPGRADE file. Remember to
upgrade the .swf file and to flush your browser cache!
Many thanks to everyone who provided feedback, patches, ideas and suggestions.
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only on the svn
trunk?
I have done something similar using a modified Flash Operator Panel
and a phone with autoanswer capabilities (polycom 501), while the
operator is using a headset. Then you can use standard manager actions
to redirect calls to the operator. Regards,
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could only make the Flash panel show that a queue 8in general)
received a call from a specific extension.
If you double click on the arrow for a queue summary button you should
see some stats, including agents logged into that queue.
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file on the flash embedded
object. Regards,
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NICE!
On 3/30/06, Joe Dennick [EMAIL PROTECTED] wrote:
I see (and like) the demo, but where can we get it?
Doug Lytle wrote:
Nicolás Gudiño wrote:
shameless plug Something like this perhaps?
http://www.asternic.org/stats/demo
It is not released
. For instance
if a call center supervisor is randomlay monitoring agent calls, and for
some reason need to intervene on a call without first having been
conferenced into the call?
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view queue stats?
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.cfg files. Please address any future question about FOP itself
to FOP mailing list (you can subscribe from http://www.asternic.org )
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want... no need to look into the Zap buttons.. and
it also shows individual timers for each person waiting on that queue.
It uses the manager redirect command
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or
unresponsive mpg123 processes.
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, etc. Anyways, FOP's
mailing list is a better place to seek for help about it, as is not
directly asterisk related.
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depending
your asterisk version.
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, but doable.
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this questions to FOP's mailing list, you can
subscribe from http://www.asternic.org or browse the archive for some
style examples. Regards,
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Local channels to find your follow-me [EMAIL PROTECTED] Like:
[myqueue]
music = default
strategy = ringall
timeout = 20
member = Local/[EMAIL PROTECTED]
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is *1.
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qualify for them). Regards,
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Does everything with AstCC work properly under Asterisk 1.2?
Yes. But checkout astcc again because it got patched to fix a bug with 1.2
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Asterisk
/view.php?id=5400
Glad to save your soul... Best regards,
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It works.
I terminated the call during the playback.
AGI debug
AGI Tx 200 result=-1 endpos=480
HUP received!
Allowing
setinuse() to get called
Please add your comments on Mantis so it gets commited quickly.
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Nicolás Gudiño
Buenos Aires - Argentina
the problem is.
http://bugs.digium.com/view.php?id=5400
Seems to fix the problem... please test and give feedback.
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writes. Regards,
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, the mailbox should probably be
[EMAIL PROTECTED] (being default the voicemail context for that extension,
as specified in voicemail.conf)
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, it will ease and make
the development faster. But the port is not easy, I will work slowly on
that. If you are a flash OOP actionscript fan and want to help, please
let me know.
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had a Polycom 600 to try. ;) Regards,
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Hello,On 8/18/05, Soner Tari [EMAIL PROTECTED] wrote:
It sounds like web_hostname in your /var/www/html/panel/op_server.cfg is setto your external ip. If you change it to your internal ip, I think you'llhave the opposite of what you describe. I couldn't find a decent solution to
this dilemma. Any
on that you should
use slightly different configurations. Contact me off list if you need
assistance.
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by Digium.
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working with it.
So my question is:
Is there any way to get FOP working together with ICD in that case?
If ICD does not use regular queues, it will require custom development
or a clever use of UserEvent in your dialplan. BTW, my rates are
cheap.. :) Regards,
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Buenos Aires
be
monitored ?
You can try by using regexp to match the channel name, prefix it with
an underscore and then a perl regexp pattern. Regards,
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this, and you must know a little javascript and/or php.
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and udp
port 5038 in iptables but without success.
Open up port 4445 tcp. Regards,
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it as a network raid-1.
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that action on the manager! (but replace ChannelA and
ChannelB to Channel1 and Channel2 as on the link event).
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, it lets you
pick up a call without loosing your internet connection, and resume
the dialup session after hangup. The only feature you need for your
telco is call waiting. It does not need forward on busy. Regards,
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I know this information can be parsed out of show channels
but I was just wondering if the is an easier way?
Its easier to use 'Status' on the manager directly. Regards,
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finished, but its progressing nicely. If you
are interested in see how does it looks like, drop me a note off list.
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reports of bad timers within FOP but I was not able to fully
reproduce the problem. You better check from the cli and compare that
with FOP display or your SNOM phone, and tell us your results.
Regards,
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-to-dial, screen pops, etc.
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, or multiple 'context' to split your buttons into
several screens. I wll send you an alternate op_style.cfg with smaller
buttons offlist. Regards,
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Hello,
I need to know the following: when a call is answered, this call es sending
a to agent, but I need that the agent when receive the call in the desktop
appear all information of this number.
You can do this with the Flash Operator Panel. http://www.asternic.org
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this,
without having to make two calls, transfer them in, then connect my self.
Maybe is not what you want, but anyways.. . you can try with the Flash
Operator Panel (http://www.asternic.org). It can do barge-ins using
meetme for that purpose. Regards,
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-Asterisk+standard+extensions
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at the license agreement
on the driver - you might not be allowed to use the software fully without
having a subscription to their services.
Florian
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around.
Drumkilla reverted the callerid changes on the latest stable (thanks
Russell!). You will be fine if you checkout stable from CVS now.
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Hello,
I deleted the config examples in the op_buttons.conf folder for how to set
up the meetme representation
[skip]
[Meetme/801]; Meetme must be defined by its room number
change the above line to:
[801]
Regards,
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handling in the
last couple of weeks. Many manager based applications stopped working
because of them. Maybe your setup is affected too. Regards,
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won't notice the
interruption. Regards,
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), you can use a dial flag (I
believe its 'p' in cvs-head?), so the recipient of the transfer will
see the original remote callerid instead of your own.
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in the RINGING event
or associated directly with the Uniqueid of the callee.
Personally I had to downgrade app_dial.c to a previous releaes to get
the callerid as before.
Just my 2 cents...
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Hello,
I've tried all the Wiki pages and still can't seem to
get this thing working and that's why I've posted this
mail.
I would like to dial two external numbers and
conference them together using the asterisk api
manager.
Hint: search the wiki for local channels
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and bridging A with B.
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. There are several kinds of cards you can purchase
them from digium at http://www.digium.com
If you do not have time to set this up, you can hire a consultant. You
can find a lot of usefull documentation, and a list of asterisk
consultants at http://www.voip-info.org
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.
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and not just anything that matches an expression in $number?
Try with:
^01154.*
being 54 the country code for Argentina and 011 the internacional prefix.
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[EMAIL PROTECTED
...
Does anyone have a good how-to?
I think you can use something like:
^ + code + .*
For example, if you want to add Argentina (54) as a route, and you
dial 011 for international:
^01154.*
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]
Copied straight from the wiki (search for fedora):
cd /usr/src (or make sure you are in your source directory)
cp configs/config-for-my-kernel .config
make oldconfig
make include/asm
make include/linux/version.h
make SUBDIRS=scripts
It worked for me last night. Regards,
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Buenos
similar problems compiling under PPC. I just removed that
module together with other troubling assembler parts (MMX detection
routines). After that it compiled fine and worked well (but the
platform is 32 bits, I dunno if it will work under a 64bit platform).
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transfers,
night mode, et all.
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to login.
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write an application that listens to the manager port and
looks for registration events and launch scripts accordingly.
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Nicolás Gudiño
Buenos Aires - Argentina
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BLF just with leds
(not LCD) Regards,
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Nicolás Gudiño
Buenos Aires - Argentina
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machine or another one).
If there is enough interest, we can make it a reality... The question
is: what features do you need and how much are you willing to pay for
it.
Regards,
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Nicolás Gudiño
Buenos Aires - Argentina
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by setting the following in zapata.conf
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
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Nicolás Gudiño
Buenos Aires - Argentina
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field set in the
CDR? What would the extension look like to record the value into the CDR?
exten = s,1,SetVar(loguserfield=${EXTEN})?
show application SetCDRUserField at the CLI
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Nicolás Gudiño
Buenos Aires - Argentina
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on the originate command.
I wonder why not?
Maybe, (just maybe, I did not try it myself) the originate variables
are passed using asterisk CVS-HEAD and variable names prefixed with
underscore... Eg: Use variable _X instead of X in the originate
command. Let me know if it works.
Regards,
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Nicolás
to
the operator.
And/Or if they did not press something it will play the message again
And/Or if they typed a wrong extension ti will read the menu again
please let me know
Thaks
Altus
http://www.voip-info.org/wiki-Asterisk+standard+extensions
Look for the t and i extensions...
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Nicolás Gudiño
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