similar to: app_queue not using exit context?

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:257 #3 0x08092b3f in handle_reload (fd=33,

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 dial the support/hotline number, get added

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

Hi, I've just uploaded a patch here: 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

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

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 back to 111. Again at this point ext 111 isn't answered it

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular

Hello, After upgrading to the latest development CVS Head, I am now getting regular errors as follows: Dec 17 17:07:30 WARNING[8092]: app_queue.c:340 changethread: Can't change device with no technology! Also, my ability to answer calls with XTen Pro softphone seems to be a bit flaky now. Any ideas? ===== Jason Goecke www.goecke.net Ph: +31.707.504.634 Mb: +31.707.504.634 Fx:

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 defining penelty for the accounts,

--- 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 we're doing

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

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

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port 9999. You can ping your broadcast IP on your network and listen with tcpdump on your network on port 9999 which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port 9999" Before I get my karma whacked again, does this work for

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

hello, I want to use randomForest to classify a matrix which is 331030?42,the last column is class signal.I use ? Memebers.rf<-randomForest(class~.,data=Memebers,proximity=TRUE,mtry=6,ntree=200) which told me" the error is matrix(0,n,n) set too elements" then I use: Memebers.rf<-randomForest(class~.,data=Memebers,importance=TRUE,proximity=TRUE) which told me"the error is

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

Here is some info that may allow some "locked" PAP2 and PAP2-NA units to be used with Asterisk: I have a PAP2-NA (from a provider other than Vonage) for which I did not know the admin password, though the "user" pages were accessible to me. The provider had set it up to fetch at startup, its configuration file by HTTP from a numeric IP. It was running 2.0.10(LSc). A search

Hello everyone, I have just noticed a fairly obvious feature that it looks like many people have been looking for... If you have a queue defined with strategy=ringall, members of the queue will still get incoming calls when they are already on a call (call waiting). The only solution that has ever worked is incominglimit=1 in sip.conf. The problem there is that it obviously disables call

I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect