Displaying 20 results from an estimated 8000 matches similar to: "Queue Announcements for Operators"
2007 Apr 24
5
tone generation
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?
If not, can I use some system command to generate the wav file
then just have asterisk play it?
Jerry
2007 Apr 25
5
Asterisk Business Edition Question
Hi,
Can anyone in the list help me with these queries on Asterisk Business
Edition.
*1. Why would anyone choose the Business Editon when the whole thing is
avalable as GPL?*
**
*2. Is there a GUI to manage asterisk?*
**
*3. Can it be compared with Asterisk NOW?*
**
*4. Is the CD a complete installation package?*
**
*5. If im looking for hiring a server on a remote location how will i be
able to
2007 Mar 19
4
Queue App - Free agent and waiting calls
<asterisk-users@lists.digium.com>Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have some
calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call in
the queue.
It happends some times.
This behavior still happend in 1.4.1 version.
Thanks a lot.
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2007 Apr 25
3
FYI
Just been getting lots of failed SIP registrations to a system here.
All coming from Taiwan.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net
Euro Tech News Blog http://eurotechnews.blogspot.com
2007 Mar 07
4
OT Vonage V-Phone Adapter (Possible Hack)
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone. Looking at the pic, it
looks like the dongle is both a soundcard and memory stick. Heck, I
would be glad to have it if I could get the soundcard to work.
Might as well since it is free after rebate.
http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem
2007 Mar 07
2
queue information in mySQL
Hi,
is it possible to have the information stored in
/var/log/asterisk/queue_log
realtime in mySQL?
thanks
2007 May 24
13
Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it production ready
for fax? By production ready I mean that it just works all the time and
doesn't need any babysitting. Do I have to worry
2008 May 06
3
asterisk queue cluster
I setup two asterisk servers with identical settings
(same extensions, same queues, etc). Each one is
connected to the same amount of incoming/outgoing
links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box).
Most extensions are sip and they register via DNS SRV
and other methods so that the two servers are load
balanced. Incoming PSTN calls (BRI) reach 50% each
server so that's load balanced
2007 Jun 03
2
Chan_mobile issue
Hello,
I just did a fresh svn install of 1.4 trunk everything. Everything
compiles and installs just fine.
When I get to asterisk-addons, I cannot select chan_mobile in
menuselect.
Chan_mobile is not even an option in menuselect for asterisk trunk.
I tried the latest patch which failed in many places but did add an
option for chan_mobile in menuselect for asterisk but it still cannot be
2007 Sep 25
2
show queue (queue name)
Hi all,
does anybody know any way that when it run "reload app_queue" in the
asterisk cli it don't lose the informations from "show queue (queue
name)" ?
I'm passing for this trouble, because I need this informations
(http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue)
that asterisk cli command "show queue (queue name)" show me
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2007 Feb 23
1
Queue Macro Problem
Hey all,
This should be an easy one. I have a few different queues and wanted to
set up a standard macro to handle them, so I can shrink the dial plan
down and stop having so much redundancy. But when I try to use it, i get
a "no answer".
Here's what does work (non macro):
exten => 5054,1,Answer()
exten => 5054,n,Ringing()
exten => 5054,n,Wait(2)
exten =>
2007 Apr 05
2
Queue call distribution
I have noticed that asterisk will only try one interface per queue at a
time. Is there any way get get it to dial say three at a time and
connect the first one that it reaches.
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2009 Dec 04
1
Get Queue values from dialplan (Was: queue_variables() function)
2009/12/4 Olivier <oza-4h07 at myamail.com>
> Hello,
>
> Has someone successfully used this QUEUE_VARIABLES() function (in
> 1.6.2-rc7) ?
> I tried to use it as I'm using SIPPEER() but without success.
>
> A previous question about it remainded unanswered (
> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
>
> Regards
>
How can
2007 May 28
5
Blindside Web Conferencing
Hello,
We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.
This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.
Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
2009 Jan 07
5
recommendation for German sound files
Hi!
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon).
thanks
klaus
2009 Jan 27
2
Module res_odbc is not loading
Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading
when I do:
module show like odbc
I have o module returned
anybody knows why?
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2009 Feb 19
2
Not answering call when queue is full or has no members
Hello all,
I'm trying to prevent answering a channel when a queue is either full or has
no members. It seems I'm forced to answer a call before I call Queue() or
else the audio is in the early media (which is unacceptable because of the
short duration of early media on ISDN).
Is there any way to let Queue() automatically answer a channel if the call is
going to be placed in the
2007 May 31
9
click to call
I have been looking around for examples or code on making a click to call
application for web sites... has anybody had any luck on this topic? Is
there any open source code out ther that could do this?
Regards
AK
2007 Apr 16
2
Problem with queue
I have queue set up in realtime on Asterisk 1.4.2.
Below is the senario that is happenening ::
I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine.
Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine.
But