similar to: Pause/UnpauseQueueMember

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, 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 =>

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, I'm trying to figure out how to do something that I hope is pretty easy. I have a remote phone system (Definity ProLogix) connected to my Asterisk system via a T1 cable (all onsite). I'd like to get some of these users on a queue hosted on the Asterisk. I've got it setup so that it seems to work OK (calls flow normally), but I'd like the users to be able to dial one

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

For all you AMP users: Why the heck is there no area in AMP to manage agents? I see you can add static agents under the queue settings, but there is no area to assign dynamic agents. Is every extension (or user) considered a dynamic agent? Is it setup up such that, as default, every extension is considered a dynamic agent that can log into any queue? (Yes, I realize you can add a password to

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.

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

Greetings and Merry Christmas, We're trying to implements a queue and agents login mechanism on our Asterisk. After going over the documentation, we're unsure if we got it right. We wish to setup a "hotdesk" mechanism, where an agent comes to a station with a PC & IP phone (that is defined as a sip friend user in sip.conf), dials a certain number (agent login extension),

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 =>

Has anyone used the H option for Queue() with Callback queues? I want customers in my queues to be able to jump out to voicemail when they get tired of waiting, but in my setup when I pretend to be a customer and press '*' [when I am waiting in the queue] I see the message 'User hit * to disconnect call.' but then just jump out to the outer loop where queued callers wait to

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

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

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

Hi, I have a question about the definition of agents. The agents.conf file looks like this: [general] persistentagents=yes [agents] maxlogintries=5 ackcall=no wrapuptime=500 musiconhold => default group = 1 agent => 1311,1311,Tom agent => 1531,1531,Tim and here is the queues.conf: [general] persistentmembers = yes [queue1] musiconhold = default strategy = rrmemory servicelevel = 60

Hi, If an agent doesn't take a call.. is there some way I can PAUSE them instead of logging them out?

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

Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122 at from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a

Hi, I have missed something so I wonder if someone could explain for me? 0424449647 desktop phone 0106024647 DECT phone 0424449630 Helsingborg queue extensions.conf --------------- [support] exten => 0424449647,hint,SIP/0424449647&SIP/0106024647 exten => 0424449647,1,Dial(SIP/0424449647&SIP/0106024647,15,rtT) [inputinterior.se] exten => 0/0424449647,1,Answer() exten =>

Greetings. I am trying to get AMP up and going on my Asterisk server. I can access the admin pages on my asterisk server via a web browser. I can add and edit things via the web browser and it edits the database accordingly. Everything seems fine except when I try to run 'amportal start'. Below is what I get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log