search for: unpausequeuememb

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

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...

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 =>

...logged on, off and on "pause". I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on "pause" using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a "dummy" extension with a hint as follows:- exten => 151,1,Dial(Agent/151) exten => 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when...

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

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

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 configured the following: exten => 6662,1,ADDQUEUEMEMBER(queue1|) ;exten => 6662,1,ADDQUEUEMEMBER(queue1|SIP/${ID${C...

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

...y 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 extension to run PauseQueueMember, and another to do UnpauseQueueMember. Is something like this possible? Answer Playback (what extension to pause) Get input <--- how do I do that? PauseQueuemember (input from user) Playback (agent paused) Hangup I have done most of this already in other contexts, but I cant figure out how to "get input" from the user...

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

...me, hold time. Also need to track an agents number of calls, number of dropped calls, refused calls, etc etc. (Also need to be able to monitor the queues for wait times, number in, number dropped, number answered, etc etc.) This seemed possible utilizing Agentcallbacklogin, and pausequeuememeber/unpausequeuemember among other commands, but not through AMP... Has anyone had luck getting this type of thing working utilizing AMP? Perhaps I should be moving on to a more robust Asterisk manager program...? Ben Ferguson CirclePix.com -------------- next part -------------- An HTML attachment was scrubbed......

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

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

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),

Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ;

...en ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding...

...en ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding...

...en ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright) * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding...

...en ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright) * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding...