You can use RemoveQueueMember(queuename) to dynamically remove the agents.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick Brown
Sent: Sunday, November 04, 2007 11:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi, All,
I have a question about agents and queues. Right now we have about 4
queues in our system. Some agents are in multiple queues. Our main
queue is for technical support and it's by far our busiest queue as
well. We have the autologoff feature set to 14 sec right now in the
agents.conf
We have a job that requires extensive knowledge of asterisk queues. The
work can be done remotely. Our customer is looking to completely
overhaul their current queue structure. Please contact me offlist if
you are interested or need more details.
- Jason
Hey Everyone,
Have a couple of questions here..
Scenario 1:
We are working with a client that currently has one support queue with
about 10 agents. They are starting to get pretty long hold times for
their customers and they have requested three queues. Queue 1 will have
a timeout of 4
Isn't that the function of an attended transfer? User3 hears User1 to
see if they want to take the call or not. User1 should then hit the
transfer key again to finalize the call.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber
Sent: Thursday, May 03, 2007 12:54 PM
If you set the queue strategy to ringall it should ring all the
interfaces you have set up in that queue. Just make sure you have
member = SIP/EXT setup.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
Novak
Sent: Thursday, April 05, 2007 4:06 PM
To:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Thomas
Sent: Friday, December 29, 2006 8:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 Random disconnects
On 12/28/06, Jason Adams [EMAIL PROTECTED
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Thomas
Sent: Friday, December 29, 2006 8:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
. It seems like asterisk gets hung up on a certain call and dumps.
Anyone else noticing anything like this?
Thanks,
Jason
Jason Adams
Sumo Systems
4694 Cemetery Road
Suite 310
Hilliard, OH 43026
Phone | 614.433.9906 ext: 102
Fax | 614.433.9931
E-mail | [EMAIL PROTECTED] blocked::mailto:[EMAIL
help would be appreciated.
Thanks,
Jason
Jason Adams
Sumo Systems
4694 Cemetery Road
Suite 310
Hilliard, OH 43026
Phone | 614.433.9906 ext: 102
Fax | 614.433.9931
E-mail | [EMAIL PROTECTED]
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this
problem when we are in a conference call with one of the employees at the
remote office and a third party. They are disconnected after a minute.
Here is the error:
Chan_sip.c 11452 do_monitor: Disconnecting call SIP/103
for lack of RTP activity in 61 seconds.
Any ideas?
Jason Adams
Sumo
I actually just deployed this server for a customer with only 4 users.
It worked great. The bios control over the IRQ's isn't the best. I
would definitely recommend against an integrated NIC. Other than that is
works great for them...
-Original Message-
From: [EMAIL PROTECTED]
Hey
Everyone,
We are in the
process of reviewing headsets for use with our GXP-2000s. I'm looking for
some feedback as to which headsets people are using, the pros and cons of those
headsets, and if they would recommend them to someone else.
Any help would be
appreciated...
-
Jason
Thanks alot for the input... I am anxiously awaiting
the release date!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corporate
IT Solutions - Michael DunneSent: Tuesday, May 09, 2006 7:31
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE:
Hello
All,
Does anyone know of
an expansion module (keypad extension for attendant) that works with the
gxp-2000?
-
Jason
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Hello
Everyone,
Here is the
scenario... I have a client who has two different companies in the same
officebut everyone works for both companies. Each person has a DID
for both companies. They only want to have one phone at their desk.
They have purchased the GXP-2000 ip phones for the
I do the same thing with outbound transfers.. Here is my code.
exten = s,3,Playback(pls-wait-connect-call)
I do this right before the dial command.
- Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Courchesne - Consultant
Sent: Wednesday,
Hey
Everyone,
I just setup a way
for our receptionist to turn on the auto attendant mode via her phone. I
setup one of the indicators to dail an extension which runs a bunch of code to
turn on/off the night mode. Is there a way using the BLF to turn the
indicator on (solid red) when night
Andrew,
Don't know if this helps your or not, but it seems like you have one too
many {} in your set statement...
You have: Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
Try: Set(DB(forward/${CALLERIDNUM}=${FORWARD}))
- Jason
-Original Message-
From: [EMAIL PROTECTED]
at all.
Hopefully someone else will give you some good advice on QOS equipment.
Joseph Tanner
On 3/7/06, Jason Adams [EMAIL PROTECTED] wrote:
Hey Everyone,
We are in the works of planning a new * installation for our company.
We have 20 users in our main office and 5 users in a remote office
Hey
Everyone,
We are in the works
of planning a new * installation for our company. We have 20 users in our
main office and 5 users in a remote office a couple of states away. Our
call volume for the main office will be anywhere from 5-10 concurrent
calls. The remote office will have
Hey
Everyone,
We are in the works
of planning a new * installation for our company. We have 20 users in our
main office and 5 users in a remote office a couple of states away. Our
call volume for the main office will be anywhere from 5-10 concurrent
calls. The remote office will have
We are using the same phones in our office with firmware 1.0.1.13 and
have no issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Tuesday, February 14, 2006 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Have you tried this:
exten = _9.,1,Set(CALLERID(num)=MAINNUMBER)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Richardson
Sent: Wednesday, February 01, 2006 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
When you dial a zap interface you have to reference the channel.
So:
exten = 190,1,Dial(ZAP/g1/800111) ; Cell Phone
Using the above, would dial out on GROUP 1. When you setup your zaptel
hardware you assign the channels to a group. You can then reference
that group and * will dial out
Richard,
This also happened to me over the weekend. What happened to me was yum
updatd two files found in /etc/udev/permissions.d/ and the other in
/etc/udev/rules.d/
Yum makes backup copies of each of these files. All you need to do is
copy the missing lines from both files and paste them
Hello
All,
We are experiencing
"heavy static" on our analog lines when dialing into our asterisk server.
We have a Digium TDM04B with 4 FXO modules. Two of the
modules are connected to our analog lines from our local
telco.
This doesn't happen
with every call. It's definitely a random
shared
with another device in your system.
On 1/3/06, Jason Adams [EMAIL PROTECTED] wrote:
Hello All,
We are experiencing heavy static on our analog lines when dialing
into our asterisk server. We have a Digium TDM04B with 4 FXO modules.
Two of the modules are connected to our analog lines
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