similar to: Feeding digit input to PauseQueueMember

Hello, Does anyone know the developer(s) of the app_queue.so application? I'm looking for the PauseQueueMember and UnpauseQueueMember features of this application for the open source version that only seem to be available on the business edition of Asterisk. Thank You, Timothy Karl tkarl@imminc.com

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

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

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Hi, I wanted to use the same extensions for Pausing and UnPausing queue members. Is that a variable that is set up with the agent status (on call, available, not logged, paused) so that I could use it to make some logic in this extension? exten => 111,1,Set(AGENTEPARADESLOGAR=${$[AGENTBYCALLERID_${CALLERIDNUM}]}) exten => 111,2,PauseQueueMember(|Agent/${AGENTEPARADESLOGAR}) exten =>

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part.... exten => *0,1,NoOp(${PQMSTATUS}) exten => *0,n,Macro(user-callerid,SKIPTTL,) exten =>

One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Hi I have a project where I need to connect an asterisk server to an avaya definity prologix PBX in order to add ip telephony to a organization, actually there's no E1 ports available for connections, I want to use a 2 port card (still dont know wich digium or sangoma), then one E1 cable will be desconnected from the PBX and connected to the asterisk server and the other port from the

Has anyone observed a problem where using Local channels with AddQueueMember results in missing TRANSFER events? Right now I'm using straight SIP channels when I call AddQueueMember(). I'm contemplating moving to Local channels because the non-state-based wrapuptime blows when you have a channel in multiple queues (they can hang up and get a call immediately so long as it's from a

I have a group of agents logged in to a queue that is set for ringall. The agents are set to auto logoff if they don't answer in 15 seconds incase they step away without logging out. That works fine, however, if they are on a call and a new call comes in, they are getting logged out too. The phones are ATA's connected via SIP. One thought is that the phones may be allowing a second

I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back.

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

Hi My name is Jose Manuel Cortes and im developing an IP telephony project, im going to interconnect a definity prologix PBX with an asterisk server (i still don't know what kind of cards i'll use digium, sangoma or voicetronix)trough a E1 connection in order to add ip telephony to the university. I would like to know if there's a compatibility problem with this. I send you a

Hi! I have a question about agents in asterisk. In first place, agent login to asterisk (from the telephone) The question is: Can an agent take a break (using a function *(some number)) from the phone? Thanks to all Regards -- Ing Francisco Roqu? 3Tech SRL Plaza Paso N?92, EP B Buenos Aires, Argentina. www.3tech.com.ar

Hi, I want to setup a pause/break between calls coming from a queue to an agent. If an agent is logged in only in one queue the wrapuptime parameter works. But what can I do if the same agent is in more then one queue? I found no parameter to configure a pause after a (incomeing) call and the next call (asterisk CVS-HEAD-04 / polycom ip500 sip phones) thanks a lot for help ciao, morel

#1 - Is there anyway to 'cancel a wrap'? That is.. when an agent is waiting their 45 seconds or so for the next call, is there anyway to cancel that and continue taking calls? #2 - I have agents who log into multiple queues. I have them staticly assigned to all queues, but they log in with their agent ids (defined in agents.conf). It seems that they will be wrapping on queue

Fellow Asterisk Users, I'm running Asterisk 1.2.5, and I've configured basic call queueing using agents. The problem I'm having is that agents who are on DND are still considered (by Asterisk) to be eligible to take calls. This means that calls will hit voicemail even when other agents who are not on DND are available. Any ideas how I can make agents ineligible to receive calls

Hi, I have a small callcenter with 3 agents who login using AgentCallbackLogin. They normally receive calls, but needs to call outside also. When they call outside, though they are busy the "show agents" shows them as available, and calls gets routed to them. How can I make them busy when they call outside. Also they also need to move out for couple of minutes or to send a mails