search for: app_queu

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:2...

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings ba...

...is still dispatched to the user if it is not in the "Not in use" status? Thanks to all in advance ************************* ************************* LOG (core debug and verbose set to 5) ************************* ************************* #grep PL1038 full [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May...

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously w...

Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent -> 2nd Agent -> 3rd Agent I've actually solved that by def...

hi everybody, Recently,I did a stress test use asterisk . when I concurrent 100 calls to the queue , i found the waiter need very a long time can listen to the music,about 40s and this time, there are about 30 waiters in the queue. everybody can tell me this reasons.whether the queue's ability is limit or the music on hold concurrent is limit . Thanks in advace! -- Best regards!

...rted music on hold, class > 'default', on channel 'SIP/incoming-00000246' > > [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:1] > NoOp("Local/mysip692 at CallFromQueue-0000003c;2", "") in new stack > [Nov 21 08:18:26] app_queue.c: Called Local/mysip692 at CallFromQueue > [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:3] > Dial("Local/mysip692 at CallFromQueue-0000003c;2", "SIP/mysip692") in > new stack > [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692 >...

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing consultative transfer usin...

...el_alloc() # 9: [0x4d270d] asterisk core_unreal.c:1222 ast_unreal_new_channels() #10: [0x4ce07e] asterisk core_local.c:1016 local_request_with_stream_topology() #11: [0x4a1fe7] asterisk channel.c:6353 request_channel() #12: [0x4a4c68] asterisk channel.c:6447 ast_request() #13: [0x7f625d021eda] app_queue.so app_queue.c:4717 ring_entry() #14: [0x7f625d0232d3] app_queue.so app_queue.c:4886 ring_one() #15: [0x7f625d025592] app_queue.so app_queue.c:5487 wait_for_answer() #16: [0x7f625d026fa3] app_queue.so app_queue.c:7110 try_calling() #17: [0x7f625d02a4e1] app_queue.so app_queue.c:8730 queue_exec...

...he applications as many as possible before submitting the code. However, I can't test all of them, so I would like to make a priority and list. If you want to use your favorite application with this feature, please reply this. I will add to the list. :) Btw, I'm considering the app_amd, app_queue for myself. :) For more detail of this feature, you can see it here. http://lists.digium.com/pipermail/asterisk-dev/2019-April/077270.html Thank you. Kind regards, Sungtae _______________________________________________ asterisk-app-dev mailing list asterisk-app-dev at lists.digium.com http:/...

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

...are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy _____ From: John Kiniston [mailto:johnkiniston at gmail.com] Sent: 28 November 2018 21:17 To: paddy at wizaner.com; Asterisk Users Mailing List - Non...

hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google&issues.asterisk.org without any clue marek

Hi all, We've been experimenting with the app_queue application, and it works quite well. The only problem we encountered was that outgoing calls (to the operators) aren't logged in CDR. Example, * operators dial a specific number/extension, and AddQueueMember(..) runs (they get added without any problems), and they Hangup. * normal users...

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing anything like that?) -JimC

...s the memebers -- not agents. queue.conf allows you to set an exit context such that if set (and you use the "T" or "t" option) allows the caller or callee to transfer the call to that context/extension. This doesn't appear to be working for me. If I set "T," app_queue looks for the extension in the caller's current context -- where the queue() call is. If I set "t," app_queue looks for the extension in the callee's context -- in my case, the context specified in sip.conf. Am I misunderstanding how this is supposed to work? Is it working f...

...re: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently ignores channel problems with its agent members. With this patch, it is easier to make statistics out of queue_log with information about problems with an agent. For example if an agent has a faulty line, or your telco/dahdi connection is having...

...sn't always tro to call this channel when people enter the queue, because of the erroneous "Ringing" state. Even when asterisk has 0 channels in use, this state can be "Ringing" for this member. This is on Asterisk 1.6.1.2. Might this be a (former?) bug in device-state or app_queue? H?kon

Hi all, does anybody know any way that when it run "reload app_queue" in the asterisk cli it don't lose the informations from "show queue (queue name)" ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command "show queue...

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member is my op. into technology SIP/operator So my queues.conf should be : [general] [default] [operatorq] music = default announce = from-queue ;context = qoutcon timeout...