search for: leaveempty

...number of callers in the queues would exceed the maxlen= value defined in queues.conf if another caller was added JOINEMPTY - a call was sent to the queue but the queue had no members, does not apply when using agentcallbacklogin since there could be unavailable members defined but not available. LEAVEEMPTY - the last agent was removed form the queue before alls calls we handled, remong callers exit with this status, also acts differently when there are only queue members that are unavaialbe JOINUNAVAIL/LEAVEUNAVAIL - same as JOINEMPTY/LEAVEEMPTY, except that there were still queue members, but all w...

...agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten => 7080,n,Hangup() exten => 7080,n(LEAVEEMPTY),Goto(ERROR) exten => 7080,n(TIMEOUT),Goto(ERROR) exten => 7080,n(JOINUNAVAIL),Goto(ERROR) exten => 7080,n(LEAVEUNAVAIL),Goto(ERROR) exten => 7080,n(JOINEMPTY),Goto(ERROR) exten => 7080,n(TIMEOUT),Goto(ERROR) exten => *210,1,AddQueueMember(teste,SIP/${CALLERID(num)}) exten =>...

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Par?ina Lama Computers Split

Hello, I am using 1.4.8 and have a question about Queues. I noticed that if I have an agent logged in using AgentCallBackLogin and that agent is unreachable for some reason (SIP phone unplugged) calls to him/her will completely yack. For example: 1-Agent 500 is the only one logged into queue number 1. 2-A call comes into queue number 1 3-The call is pushed to agent 500 at extension 21 which is

...empted unlock mutex 'peer' without owning it! (Reported by Alec Davis) * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-28001 - res_pj...

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>

hi list, i have a question about the args of queue: when we use Queue() app, there are some arguments than can use. help from CLI: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) well.. i'm trying to identify who has taken the call on a queue, and, when agent conected, launch a macro with some args based on who takes the call i do: exten =>

...* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27398 - No joint capabilities with video and audio-only streams (Reported by Benjamin Keith Ford) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-27999 - Wrong SRTP use status report (Reported by Salah Ahmed) * ASTERISK-28001 - res_pjsip_registrar: Improve...

Hey Everyone, Have a couple of questions here.. Scenario 1: We are working with a client that currently has one support queue with about 10 agents. They are starting to get pretty long hold times for their customers and they have requested three queues. Queue 1 will have a timeout of 4 minutes. After that it will move to Queue 2 which has a default timeout of 3 minutes. After that we will

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

Hi, Is there any way to run a command (AGI script, whatever else) at that moment when the call that was in the queue is being bridged to a specific agent? An examples of what I would want to do with this is, for example, have Asterisk ask the caller for his 4 digit customer ID before being put in the Queue. Once I know who the caller is being connected to (which agent) I'd run a

...* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27398 - No joint capabilities with video and audio-only streams (Reported by Benjamin Keith Ford) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-27999 - Wrong SRTP use status report (Reported by Salah Ahmed) * ASTERISK-28001 - res_pjsip_registrar: Improve...

...* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27398 - No joint capabilities with video and audio-only streams (Reported by Benjamin Keith Ford) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-27999 - Wrong SRTP use status report (Reported by Salah Ahmed) * ASTERISK-28001 - res_pjsip_registrar: Improve...

...* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27398 - No joint capabilities with video and audio-only streams (Reported by Benjamin Keith Ford) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-27999 - Wrong SRTP use status report (Reported by Salah Ahmed) * ASTERISK-28001 - res_pjsip_registrar: Improve...

...27398 <https://issues.asterisk.org/jira/browse/ASTERISK-27398>] - No joint capabilities with video and audio-only streams (Reported by Benjamin Keith Ford) - [ASTERISK-27973 <https://issues.asterisk.org/jira/browse/ASTERISK-27973>] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) - [ASTERISK-27997 <https://issues.asterisk.org/jira/browse/ASTERISK-27997>] - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) - [ASTERISK-27999 <https://issues.asterisk.org/jira/browse/ASTERISK-2799...