similar to: Asterisk in the call center - how do you do it?

Hi, I'm having hard time to emulate agencallbacklogin. Agent can logon and receive call without any problem using addqueuemember. The problem comes when I try to evaluate their performance using queuemetrics. Here is an exemple of my log script: ;Agent Login exten => _60XXX,1,Macro(agentLogin) [macro-agentlogin] exten => standard,1,AddQueueMember(queue1) exten =>

Copying to asterisk-users, as it's of use there too. I copied this code years ago from the net, it may have been modified since... This however is only used by managers, as it allows the manager to log a user in and out. For agent logged in/out status: where 8501 is the queue number and 8512 is the agent's extension, and SIP0001 is the agent's device. in extensions.conf

greetings > > I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot > > understand what the difference between a queue member and queue agent is. > Agents would be people who's job it is to answer calls. An agent logs in=20 > indicating that he's now available to take calls. Asterisk then sends calls= > to each agent as they are free to take a

greetings - I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot understand what the difference between a queue member and queue agent is. Gurus, can you please explain this? When - for example - should I use "AddQueueMember" application and when should I use "AgentLogin" ? Respectfully Constantine

On Mar 19, 2008, at 1:00 PM, asterisk-users-request at lists.digium.com wrote: > Am I expecting too much? Perhaps. I think the hardware on which we run Asterisk can be much more reliable than the software, which is often the case. We have a bunch of HP servers with RAID and have never lost anything. A HD may fail, but the RAID keeps it going until we pop a new drive in there. A

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy "fixed" the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) After doing this, I recompiled ztdummy and it worked. Note

>> All too common and largely undocumented. I had this same problem. >> >> Installing ztdummy changes Asterisk to use it for timing of playback, >> apparently. Removing ztdummy "fixed" the problem. To get it all to >> work, I had to upgrade to to at least kernel 2.6.23.11 (previous >> versions are either missing options are just broken.) > >

and I found a reference to an AgentLogin.rtf. Looks great, except I can't get it to work. queues.conf: [sjs-testq] music = default timeout = 1 retry = 1 maxlen = 0 member => Agent/10001 agents.conf: agent => 10001,1234,Steve Sobol extensions.conf: (I have a phone line set up on which the main menu tells you to press 1 to be added to queue. Pressing 1 lands you here) exten =>

Hi David, I've got probably the same/a similar problem. Do you add the phones to the queue (AgentLogin/AddQueueMember)? If there are entries like: " Spawn extension (macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have the same problem like me. I suspect that something goes wrong with the nested macro calls within the AMP-generated dialplan, so what I

Hello all. Anybody around that is utilizing the PauseQueueMember and UnpauseQueueMember applications? Or even the AddQueueMember and RemoveQueueMember applications? I'm trying to set these applications up to function in relation to the agent number, rather than the extension the agent is at. I'm not having much luck. Anybody have any pointers or suggestions on how to get these

http://bugs.digium.com/bug_view_page.php?bug_id=0001693 This patch adds a lot of options for AgentLogin/AgentCallbackLogin Please test and respond in the bug tracker! /O ------------------------------------------------------------------------------------- "This patch adds quite a few new features into __login_exec () of channels/chan_agent.c for AgentLogin() and AgentCallbackLogin(). Only

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some

On Oct 16, 2008, at 2:36 AM, asterisk-users-request at lists.digium.com wrote: > I want to call an extension like 88888 and invoke an external C > program upon > calling, pass an constant integer like 1 to the C program. > > What I have done is: > > /etc/extensions.conf: > exten => 88888,1,system(/usr/local/src/parallel/fire 1) > exten => 88888,n,

I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have configured queues.conf ---------------------------------- [irock.com] strategy=leastrecent

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24