similar to: Looking for real world uses of CallbackAgentLogin()

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 =>

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Emrah wrote: >> This is an asterisk-users question, and would have been more appropriate to have >> asked there. >> >> Instead of setting up your conferences in meetme.conf, you could set them up >> dynamically in the dialplan, and then you can control whether the user is >> prompted for a pin or not when joining the conference, based on whatever logic

Hello all. I'm setting up a new call center PBX using * v1.4, and figure it's better to go with AddQueueMember over AgentCallbackLogin. The functionality of AgentCallbackLogin still works, but without a firm idea of how long it will be in the codebase, I'm wary of building a system on top of it. The basic mechanics work, but I'm having some trouble on logging. With

Hi, I'm currently migrating to 1.4 and have problems changing deprecated AgentCallbackLogin to AddQueueMember. I have dynamic queues and dynamic agents (MySQL Realtime), and pseudo-dynamic agents.conf (with huge amount of possible agent numbers). Agent login is done trough manager API: * AgentCallbackLogin * QueueAdd In 1.4 seems AddQueueMember can do all the same, but there is no such

I am developing an ACD front end using Asterisk 1.2.14. I heard that AgentCallBackLogin will be deprecated in future version of *. Is this true? If it is, how can I use AddQueueMember to replace AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have multiple queues and a lot of agents defined in queues.conf and agents.conf. Each agent may login more than one queue. It

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes

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:

Hello users! Recently I read that AgentCallbackLogin is going to be deprecated soon. Wanting to set up a few callback type queues, I set them up as suggested in queues-with-callback-members.txt. I was able to set the queues up completely this way, however, I'm trying to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login status. FOP monitors their status if I call

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

Hi, I'm confused with agents and queues in Asterisk. If I use AddQueueMember() then "show queues" shows the agents that I have logged into the queue... however the agent ID has to be the extension the agent is sitting at ... kinda useless for stats tracking. If I use AgentCallbackLogin() then "show queues" shows no agents logged in, but it works and "show

Hi Mark, Are you still having audio problems between outside SIP channels? Make sure that you have set the following for all SIP channels in your sip.conf canreinvite=no -- sudhir > Message: 2 > Date: Mon, 26 Jul 2004 22:46:22 -0400 > From: Leif Madsen <leif.madsen@gmail.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Nat...again.... >

Thanks for your reply. canreinvite has been set to "no" from the beginning...still no luck. Maybe I'll be able to take a trace of it tonight...we'll see...but any thoughts at all are appreciated! -Mark > > Hi Mark, > > Are you still having audio problems between outside SIP channels? Make > sure that you have set the following for all SIP channels in your

Hi Leif, Thanks for the information. I checked the /tmp/ folder and there was core #### files and I tried to back trace it but it was not showing the cause of that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from past few days its going on fine. I have also researched and found that version 1.4.17/18.1 had the issue of channel stuck up as well as random asterisk crashes.

Hi, I've been fussing with this for a while now - and cannot seem to get it to work correctly - or rather as I desire it.. I'm trying to implement a 'find-me' feature in my voicemail. Basically - pressing 1 at the voicemail puts the caller into a queue. The members of the queue are my cell phone and my work DID.. I need the queue to call both numbers at the same time and wait

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

Good morning?... I'm?using?Asterisk?1.4.40?AgentCallbackLogin?in?a?Call Center.?What?is?happening?isthat?when the?Call?Center?has more?than 15?simultaneous calls?the?login application?isextremely?slow to?fall into?the?low?priority,?ie, the?agent?can?log in,?but?takes about 1minute to?drop?in?priority?below.?.. I've tried?to recompile?the?asterisk, I