the base code?
>
> Paddy
>
> --
> *From:* Leon Wright [mailto:lwri...@corpcloud.com.au]
> *Sent:* 30 November 2018 02:17
> *To:* pa...@wizaner.com; asterisk-users@lists.digium.com
> *Cc:* johnkinis...@gmail.com
> *Subject:* Re: [asterisk-users] Queues and penalties
>
> Padd
...@wizaner.com; asterisk-users@lists.digium.com
Cc: johnkinis...@gmail.com
Subject: Re: [asterisk-users] Queues and penalties
Paddy,
This appears to be how the queue app works. I ended up patching the queue
app:
diff --git a/apps/app_queue.c b/apps/app_queue.c
index e3a4e22..72072d0 100644
--- a/apps
shortcoming in app_queue.
>
> Any ideas, suggestions, anyone want to work with me to sort this ?
>
> Paddy
>
>
> --
> *From:* John Kiniston [mailto:johnkinis...@gmail.com]
> *Sent:* 28 November 2018 21:17
> *To:* pa...@wizaner.com;
28 November 2018 21:17
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Queues and penalties
This should work, How are you defining your timeouts in the queues.conf ?
And to verify, in your extensions.conf you are calling Queue with the queue
This should work, How are you defining your timeouts in the queues.conf ?
And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf?
On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote:
> Hi All
>
> I have been looking at this
Hi, what I did, I mixed the music on hold to have the announce in at a
specific time without leaving queue
On 25 February 2016 at 16:53, Daniel Chavez wrote:
> Ish,
> I use the same version of Asterisk on CentOS 6.7. I wonder the same thing.
> Hopefully we will find this
Ish,
I use the same version of Asterisk on CentOS 6.7. I wonder the same thing.
Hopefully we will find this out.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Martin amar...@xes-inc.com wrote:
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 28, 2015 12:12:05 PM
Subject: Re: [asterisk-users] Queues don't
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 29, 2015 11:53:13 AM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 28, 2015 12:12:05 PM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members
In your queues.conf do you have a leavewhenempty and joinempty set?
in queues.conf
[myqueue]
leavewhenempty = strict
joinempty = strict
strategy = ringall
ringinuse = no
On Tue, Jul 28, 2015 at 9:58 AM, Andrew Martin amar...@xes-inc.com wrote:
Hello,
I am running Asterisk 11 on CentOS 6.x.
You need to do this when the call connects. If you can do this within a
couple of seconds, this is usually good enough to be usable (that's what
we do on the QueueMetrics agents pages).
Thanks
l.
2013/8/3 Timothy Smith timotsm...@gmail.com
Hello Folks,
I am setting up a call center but we
Hi,
Our queue members are Local channels, thus when dialing the agent, the
dialplan will do several stuff including:
Set(CALLERID(name)=${CALLERID(name)}:Sales)
UserEvent(something,data: ${bunch-of-data-in-some-format})
Dial(SIP/final-agent-phone,timeout,A(Sales))
The UserEvent will be picked
Dear Mitch,
Thank you so much. This partly solves my problem by a great deal, as
we'll send a message to the agent immediately on picking the call. As
the agents are local SIP channels, I will attempt looking up the
caller's name (if it exists in our database) and set it prior to
entering the
Dear Tiago,
Thanks for your answer, but I have a few questions.
Do you use queues? We are operating a call centre with several queues,
so I don't see how we would use the Dial command. When a call comes
in, we enter the caller (depending on what options he has selected)
into a queue. Do you have
Hi,
You just said you use Local channels. Local channel is a dialplan that has
a Dial() to a sip device?
We use queues, and have a queue-macro that sends the UserEvent upon
bridging the call...
On 4 August 2013 16:41, Timothy Smith timotsm...@gmail.com wrote:
Dear Tiago,
Thanks for your
We do something very similar.
Use the gosub parameter of the Queue application to call a subroutine in
the dial plan when the agent answers the call.
same =n,Queue(sales,tc,,sub-QueueConnected)
[sub-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
; whatever
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is connected (ie registered) to server A
-
2013/1/25 Alec Davis siva...@paradise.net.nz
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is
I've not tried to publish device state with XMPP yet but I've
discovered this issue
https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as
one asterisk server so hopefully, I won't be hit by the issue
above but have you met
On 01/25/2013 01:59 PM, Alec Davis wrote:
I've not tried to publish device state with XMPP yet but I've
discovered this issue
https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as
one asterisk server so hopefully, I won't be
Not that this is an excuse or a valid workaround for
everyone, but I believe that issue won't apply if you're
using Asterisk 11 and res_xmpp.
res_jabber: yup, totally still a problem.
Hmm. We're using Asterisk 11, but I still think res_jabber.
Why havn't I changed to res_xmpp, I have no
On 26/11/2012 10:14 AM, Klaverstyn, David C wrote:
Hi All,
I’m new to Queues and I have created one as follows which seems to work ok.
[david-test]
strategy = rrmemory
timeout = 10
retry = 0
maxlen = 0
announce-frequency = 0
announce-holdtime = no
member = SIP/121
member = SIP/122
2011/5/5 Olivier oza_4...@yahoo.fr
Hi,
If my memory serves me right, up to Asterisk 1.6, Queue app internals kept
the application from working some other apps such as PickUp.
I wonder if such things are possible (and if possible, still keep useful
Queue Logs ie logs in which picked up or
The Queue() application can automatically pause members who fail to
answer; this would be the solution to your problem. With that solution
in place, though, the agent will still need to be able to un-pause
when they return to their desk, and since that is the case, they
really should be
I am a little confused as to what the OP wants the system to do? Call the
proper agent, but when they don't answer, on the next call, it shouldn't
call the same agent? OK, but for how long? 5 minutes? Until they manually
unpause (current option as described by Kevin), 30 minutes? Should it
Mike wrote:
I was hoping to use this Queue not for professional agents in a call center,
but for reception. When the receptionist (lowest penalty) is not at the
desk, then some junior sales person can pick up those calls.
We have our receptionist setup in a front-desk queue that has 2
We have our receptionist setup in a front-desk queue that has 2 phones in
it.
The incoming call rings directly to the phone for 30 seconds, if not
answered, plays the, Please wait while we find someone and then drops
them into a queue. At this point, it rings the operator phone again and
On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote:
The Queue() application can automatically pause members who fail to answer;
this would be the solution to your problem. With that solution in place,
though, the agent will still need to be able to un-pause when they return to
their
Barry,
I'm using the Asterisk GUI. When defining a User extension (menu 'user')
the only option I have is Is Agent.
The SIP extension (11) is automatically created as an agent that needs
to log in.
In the advanced options I can manually edit queues.conf and change to
member=SIP/11.
This way of
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
jonas kellens wrote:
Is it possible to make use of queues for incoming calls but to have
agents that do not need to log in ?
Make the phones members of the queue. In queues.conf:
[MY_QUEUE]
member = SIP/1234
member = SIP/5678
etc.
Barry
Simply use
member=SIP/Tarek
member=IAX2/JONAS
member=LOCAL/whatever
simple and good..
with member=SIP/extension i'm facing a CALL WAITING issue.. the agent hears a
callwaiting signal whenever the queue tries to call .. so i woul dsuggest using
call-limit and busy limite with all your Agents
It should be realistic, but have you considered just using followme to add
the cell phones to the queue list?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis
Elsberry
Sent: Monday, November 16, 2009 3:25 PM
To:
manually each day. Did I
overlook something in how followme works?
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 1:37:04 PM
Subject: Re: [asterisk-users
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues
I had looked at followme as a solution but ran into the same stumbling block
of having to hard code the cell phone list. I didn't see a dynamic way of
the list being extensions 12 and 14 on Monday, but changing to extensions 13
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Queues
Since followme is extension-based, you have at least two options. Option 1
is to have a few extensions designated for following where you punch in the
cell numbers as you wish. Option 2 is to use
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 2:23:49 PM
Subject: Re: [asterisk-users] Queues
Hi Travis,
There’s lots of different ways to attack “on-call” roster solutions in Asterisk
– as Danny suggested, FollowMe() is definitely an option
] On Behalf Of Travis Elsberry
Sent: Tuesday, 17 November 2009 10:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues
Hi Michael,
Your web interface for the on-call roster is pretty close to what we're
trying to trying to achieve. I would like
Rilawich Ango escribió:
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the autopause works after member failed to answer call.
However, other queues don't work for the autopause function.
queue 1000:
-- Nobody picked up in 25000
Thanks. Finally, I find that it was caused by the use of the table wrongly.
On Thu, Oct 22, 2009 at 10:23 PM, Miguel Molina
mmol...@millenium.com.co wrote:
Rilawich Ango escribió:
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the
C. Chad Wallace cwall...@lodgingcompany.com writes:
OK, I decided to write it up in AEL. It's incomplete and untested, but
it probably gets the idea across a little better.
context agentcalls {
_2XX = {
Set(AGENT=${EXTEN}); // Assuming agent ID is extension.
if
Benny Amorsen benny+use...@amorsen.dk writes:
Would it perhaps work to simply Wait(30) if the call is rejected by the
phone? If the Queue assumes that the phone is busy for those 30 seconds,
I have accomplished my goal. It's worth a shot.
This works! Actually I tried out Wait(1000), but that
At 11:23 AM on 16 Oct 2009, Benny Amorsen wrote:
I was going in the same direction at the end of my first mail, but I
hadn't written any code. There is a problem though: The Queue
application will keep sending calls to the Local channel, which have
to be rejected, over and over.
Would it
C. Chad Wallace cwall...@lodgingcompany.com writes:
It would only be trying one agent at a time for each waiting queue
member...
Would it? Almost all our queues are on a ringall strategy.
I don't know how expensive it is to open and close a Local channel and
do a DB lookup, but I wouldn't
At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote:
C. Chad Wallace cwall...@lodgingcompany.com writes:
Also, if there is another agent available, the caller would be
connected immediately, and it wouldn't have to make any more
attempts. With the Wait() solution, that caller would be
Elliot Otchet elliot.otc...@callingcircles.com writes:
Have you tried autopause=yes in your queue configuration? You can then
unpause the member by either the dialplan (e.g. having the cell phone
user log back in) or using an AMI based program to change the
paused state.
You can read more
Lenz Emilitri lenz.lo...@gmail.com writes:
You could configure them as agents and have them log off automatically
after a while they're not responding.
Agents have to log in and wait for calls though, don't they? There used
to be AgentCallbackLogin, but that has been replaced by dialplan code
That shouldn't be too hard to accomplish. If you've got the addons (and mysql)
installed you could store them in a MySQL table (timestamp, device) and have a
cron job set to run at X frequency that un-pauses the queue members via AMI.
Don't want to go to MySQL? Use system() to 'touch' files
Elliot Otchet elliot.otc...@callingcircles.com writes:
That shouldn't be too hard to accomplish. If you've got the addons
(and mysql) installed you could store them in a MySQL table
(timestamp, device) and have a cron job set to run at X frequency that
un-pauses the queue members via AMI.
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending the next call.
Queues.conf has a million settings, but I can't
At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote:
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending the
You could configure them as agents and have them log off automatically after
a while they're not responding.
l.
2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP
What is the command to log off the agents ?
Thx
On Wed, Oct 14, 2009 at 6:45 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
You could configure them as agents and have them log off automatically
after a while they're not responding.
l.
2009/10/14 Benny Amorsen
Have you tried autopause=yes in your queue configuration? You can then unpause
the member by either the dialplan (e.g. having the cell phone user log back
in) or using an AMI based program to change the paused state.
You can read more about the latter here:
Take a look at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random
You should be able to do what you want with this, it obviously won't take in
to account the actual amount of people still in the queue (for example if
someone hangs up while on hold). I'm sure there'd be a way of
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup
exten = 123,10,Queue(queue_2)
exten = 123,11,Hangup
Joao Pereira
--
StarTel - A Rede Livre
Joao
On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup
exten = 123,10,Queue(queue_2)
exten =
Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] queues load balancing
On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10
Danny Nicholas schrieb:
Here is a brute force solution:
[global]
CALLCOUNT=0
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
...,Set(CALLCOUNT=$[${CALLCOUNT} + 1])
or
...,Set(CALLCOUNT=${MATH(${CALLCOUNT}+1,int)})
exten =
Hi,
My apologies Nicolas, a mistake from my part. And I appreciate for
correcting me. Asternic is a good piece of work.
Regards,
Kurian Thayil.
On Mon, 2009-07-06 at 09:41 -0300, Nicolás Gudiño wrote:
Hello,
Just a correction, Asternic Call Center Stats is not from
asteriskguru.
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,
--
Nicolás Gudiño
Hi Sriram,
1. Set the channel variable MonitorFilename before Queue() in dialplan
and you can give some meaningful filename for record.
2. I guess you can use an AGI to capture events and then integrate this
with a DB in the Backend. This should help you to track the activity.
3. asternic from
Gabriel Ortiz Lour wrote:
Hi all,
After * starts the command queue show would not show any of the
realtime queues, but just the ones that are in the queues.conf file. In
this state de AMI would not send any QueueMemberStatus for that queues
until a call is received by that realtime
Mark Michelson escribió:
Gabriel Ortiz Lour wrote:
Hi all,
After * starts the command queue show would not show any of the
realtime queues, but just the ones that are in the queues.conf file. In
this state de AMI would not send any QueueMemberStatus for that queues
until a call is
The most popular answer I've seen here is to replace the regular music with
a streamed audio feed which can be anything you have access to. I'd try and
give you details, but they wouldn't be correct. This information is pretty
easy to locate in the digium site, viop-info.org or google.
- Cary Fitch ca...@usawide.net wrote:
I am trying to get a queue to do more than just play music and hold
calls.
Specifically, making some comforting voice announcements would be
nice.
You may want to take the quotes off of the filenames in your queues.conf config
file... they're not
On 3/20/09, Cary Fitch ca...@usawide.net wrote:
I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some comforting voice announcements would be nice.
Below is the queues.conf file relevant portions.
Member phone number is munged to protect the
hi
we ALWAYS use
sip phone IP*-E1-pstn
or
sip phone-IP-*-E1-legasypbx-?- pstn
we use the digiums cards whit echo canceller and we havent any echo problem.
more than 20 or 30 installations.
whit almost every provider in the country.
dont be SO scared.
David
2009/1/26 Sriram
Benoit wrote:
Hi,
I'm a little surprised, up until 1.4.22 my agents where using an IAX
channel to ZoIPer Softphone,
however since after the upgrade to .22 we experienced a problem with
hangup failure between zoiper
and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i
Mark Michelson a écrit :
Benoit wrote:
Hi,
I'm a little surprised, up until 1.4.22 my agents where using an IAX
channel to ZoIPer Softphone,
however since after the upgrade to .22 we experienced a problem with
hangup failure between zoiper
and asterisk (look like bug
Thomas Winter wrote:
Hi,
iam using and queue and starting an AGI script after caller connected to
agent.
How to find out in the script the connected agent, MEMBERINTERFACE seemed to
be not work, either as variable in the queue command and also not in the AGI
script.
How to found out
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
Is there any way to modify the volume (either lower the volume of the
clients, or increase the volume
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
Is there any way to modify the volume (either lower the volume of the
clients, or increase the volume
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
Is there any way to modify the volume (either lower the volume of the
clients, or increase the volume of the agents) while doing the join of the
-in and -out files into one recording?
Uh-huh. Read the documentation for
-Ursprüngliche Nachricht-
Von: Rob Schall [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 9. April 2008 15:50
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Queues +Exiting
I'm having a problem getting my queue to function as it should.
That fixed it. I always thought the s would be the fall back from all
extensions that didn't match. I guess that doesn't work in this case.
Thanks!
Rob
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Rob Schall [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 9. April 2008 15:50
On Wed, Oct 31, 2007 at 06:11:47PM +0100, Louis-David Mitterrand wrote:
Hi,
Using 1.4.13 is it possible to ignore 302 redirects from sip devices
belonging to a queue?
For a queue that rings the whole office it doesn't seem very useful to
obey a redirect programmed on a phone.
It
Tim Groeneveld wrote:
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote:
When users call 510 then, it actually does ring everyone who has called
511.
The problem is when the operator comes to pick up the call. The operator
hears nothing, and the user still hears the Music on
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to
On Monday 20 August 2007 8:16:32 pm Atis wrote:
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
Can you also provide output of show queues and show channels? Plus
the logfile of dial to 511.
I'm using QueueAdd after AgentCallbackLogin (trough manager API).
Maybe you need to use
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
On Monday 20 August 2007 8:16:32 pm Atis wrote:
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
Can you also provide output of show queues and show channels? Plus
the logfile of dial to 511.
I'm using QueueAdd after
Tim Groeneveld wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote:
When users call 510 then, it actually does ring everyone who has called
511.
The problem is when the operator comes to pick up the call. The operator
hears nothing, and the user still hears the Music on Hold. Not only that,
On 7/30/07, voiplist [EMAIL PROTECTED] wrote:
I noticed that if I have an agent logged in using AgentCallBackLogin
and that agent is unreachable for some reason (SIP phone unplugged)
calls to him/her will completely yack.
For example:
1-Agent 500 is the only one logged into queue number 1.
, 27 de Julho de 2007 12:55
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Queues strategy leastrecent
Dnia 2007-07-27, o godz. 11:09:37
Marco Campos [EMAIL PROTECTED] napisał(a):
I think this strategy should work like this:
a) Make a list of available
Dnia 2007-07-27, o godz. 11:09:37
Marco Campos [EMAIL PROTECTED] napisał(a):
I think this strategy should work like this:
a) Make a list of available agents and their idle time (time
since last call) and update it each time a call gets answered or ends
b) When a
You might want to have a look at QueueMetrics: http://queuemetrics.loway.it/
I am not sure if it supports all features you are looking for but it
should be a good start.
=Stefan
--
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221
On 7/12/07, Stefan Reuter [EMAIL PROTECTED] wrote:
You might want to have a look at QueueMetrics:
http://queuemetrics.loway.it/
I am not sure if it supports all features you are looking for but it
should be a good start.
QueueMetrics is working well for me in a 75 seat call center, but it
Hello Voipcrazy,
It's funny you should mention that - we've just released (as in today) a FREE
version of our OrderlyStats service for call centre and queue monitoring and
management.
OrderlyStats features realtime (synchronous/message-based) display of all
queue, agent and caller events so
Ciao Andrea,
Hello *,
do queues allow me to set an announce like the A() option of the Dial() cmd?
The announce that I've found is a message that is heard by the caller. I'd
like to send a message to the member of the queue that picks up the call.
In order to help people that find this
Why don't you make up the MOH in order to play your sound files, as you
need?
l.
On Mon, 07 May 2007 16:29:28 +0200, Andre Courchesne - Consultant
[EMAIL PROTECTED] wrote:
Hi,
Anyone knows if there is a way to play a list of sound file in a
round robin mode (at specific interval)
It is indeed a way, but not very flexible for my customer that want to change
the sound files weekly or even daily.
I found this patch that might do the trick:
http://bugs.digium.com/view.php?id=6681
Need to test it. So far I could not find a way to do so without patching
asterisk.
You can have the agent login once and newer log out. You can certainly set up
your asterisk box to persit the login over the reload and the restart.
persistentagents=yes
Regards,
Sanjay Rajdev
Phone : +1 (877) 342 2329 x 1702
Fax : +1 (815) 261 5907
http://www.featherstoneinformatics.com
Is there anyway to setup a queue with only one agent (device) which is
always logged in. So when a call hits that queue the device will ring (if
not already on a call) or will be put in the queue if the call is already
in place?
Sure, in queues.conf you can add many type of
Yes.
On Sat, 17 Mar 2007, Steve Kennedy wrote:
A quick question on queues in Asterisk, if you specify a specific
resource as a queue member (i.e. member = SIP/40 say) is it
automatically a member of the queue without having to specifically log
on via AgentLogin stuff?
I under stand if you
On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote:
Yes.
to which bit? auto-agent (as per resource)
or voicemail to an agent?
Steve
On Sat, 17 Mar 2007, Steve Kennedy wrote:
A quick question on queues in Asterisk, if you specify a specific
resource as a queue member (i.e.
If you make a SIP device a queue member, that member will be rung so
long as the device state of the SIP interface shows as not in use.
With regard to voicemail, are you trying to get a queue call answered
by voicemail or is that not your intent?
On 3/17/07, Steve Kennedy [EMAIL PROTECTED]
On 2/15/07, Angel Heart [EMAIL PROTECTED] wrote:
cud any one help me figuring out the problem... When the agent in a queue is
engaged, it cannot accept anymore calls, below is the script;
Angel,
Check your queues.conf, specifically the joinempty parameter.
See below the relevant part in
Hi Ex vito,
Thank you for your response below is my current config. I defined it into my
queues_addintional.conf where the definitions of queues defined. Do I need to
defined it in the general portion of queues.conf? But anyway, there's no harm
for trying.
I am using Asterisk 1.2.13 svn rev
Am Friday 02 February 2007 23:48 schrieb BJ Weschke:
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote:
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not
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