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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, 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
Regards,
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-to-dial, screen pops, etc.
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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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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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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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and bridging A with B.
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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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Nicolás Gudiño
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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), 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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won't notice the
interruption. 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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around.
Drumkilla reverted the callerid changes on the latest stable (thanks
Russell!). You will be fine if you checkout stable from CVS now.
Regards,
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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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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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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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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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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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by setting the following in zapata.conf
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
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BLF just with leds
(not LCD) Regards,
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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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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).
Regards,
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...
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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Nicolás Gudiño
Buenos
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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.
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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
Good luck,
--
Nicolás
that means polling the server periodically, comsuming resources,
instead of being notified just once.
Latest CVS does include manager events for register / unregister /
reachable / unreachable / lagged events for sip and iax peers. You
will still need to poll for initial status.
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http://bugs.digium.com/bug_view_page.php?bug_id=0002460
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.. Best regards,
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that will give better sound from compressed
formats?
Thanks
I use this script to convert to .gsm with great success:
#!/bin/bash
BASE=`echo $1 | cut -d\. -f1`
echo $BASE
sox $1 -t gsm -r 8000 -g -b -c 1 $BASE.gsm resample -ql
Best regards,
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1000 in 'conferences'
context.
This feature is already implemented and working in the next to be
released version of the flash panel (but now it will only dial numbers
predefined in the panel itself). You can get it from
http://www.asternic.org
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with it..
Regards,
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sugestion is welcome..
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|Mp|
My extension is 3044 and I am NOT on any calls right now. Somehow this
didn't get destroyed when I hung up. Rebooting my phone didn't change
anything.
How can I get rid of those?
Try with:
soft hangup CHANNEL
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, but they
didn't seem interested.
They would have to change their brand name to IAXURA
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. Maybe you have to
look at those java wrappers that mix agi with manager.
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. Is this possible?
I have just implemented this in the Flash Operator Panel, I use:
Action: Command
Command: show agents
And then some perl magic... FYI, every CLI command can be executed
from manager this way. Regards,
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regards,
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configured for
; standard U.S. tones
You might also want to take a look at application BackgroundDetect, just issue:
show application BackgroundDetect
from the CLI
Good luck,
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in agents, and it shows some statistics on queues and
individual agents also.
http://www.asternic.org
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, or the same pitch but a really slow speed for some
recordings.
Best regards,
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first call to the
agentlogin extension number. After login in, any call that arrives to
a queue that agent is a member of, it will pick up the call without
user intervention. If there is no call, they will hear music on hold
till the next call.
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the sipura faq
http://www.sipura.com/Documents/faq/Section_2.html#5
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Hi,
On Tue, 26 Oct 2004 14:01:14 -0500, Matthew Boehm [EMAIL PROTECTED] wrote:
-r and -R are the same thing except -R tries to reconnect if dropped.
regarless, -r didn't work either.
su root and try again...
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?
Asterisk detects hangups with busydetect and busycount just fine. At
least for me. The problem is ANSWER detection for billing purposes.
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in the queue LID
Help!
You will find your answer in FOP Mailing list. Regards
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= 1,4,Goto(queues,1,6)
exten = 1,5,SetVar(TIMEOUT=900)
exten = 1,6,Queue(support,t,,,${TIMEOUT})
exten = 1,7,Voicemail([EMAIL PROTECTED]);
exten = 1,8,Hangup
Regards,
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[EMAIL
possible. Then the call will follow
the dial plan if I have
forwarded my number to another extension.
I think it will work. Try with:
Local/[EMAIL PROTECTED]
You can put that as the Channel parameter in the manager originate command.
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looking links to
forged sites. You cannot trust email links either.
Regards,
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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
, 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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or
unresponsive mpg123 processes.
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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
Best regards,
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native attendant transfers with asterisk 1.6.
Best regards,
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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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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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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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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
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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listening to the events.
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on this interfaces should be usability and flexibility
and not only looks.
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they've stopped working.
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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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can join the Meetme room muted...
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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
Best regards,
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use slightly different configurations. Contact me off list if you need
assistance.
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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
had a Polycom 600 to try. ;) Regards,
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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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that action on the manager! (but replace ChannelA and
ChannelB to Channel1 and Channel2 as on the link event).
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it as a network raid-1.
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and udp
port 5038 in iptables but without success.
Open up port 4445 tcp. Regards,
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this, and you must know a little javascript and/or php.
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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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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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by Digium.
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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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.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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. 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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Buenos Aires - Argentina
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link, etc.
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Nicolás Gudiño
Buenos Aires - Argentina
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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
file on the flash embedded
object. Regards,
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Nicolás Gudiño
Buenos Aires - Argentina
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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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Nicolás Gudiño
Buenos Aires - Argentina
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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Nicolás Gudiño
Buenos
, 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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Nicolás Gudiño
Buenos Aires - Argentina
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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Nicolás Gudiño
Buenos Aires
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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Nicolás Gudiño
Buenos Aires - Argentina
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Nicolás Gudiño
Buenos Aires - Argentina
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