similar to: GROUP_COUNT not counting correctly

I have a macro and when I call it I have something like this: exten => s,1,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) exten => s,n,Set(GROUP()=MYGROUP) ;Set Group exten => s,n,NoOp(Group List: ${GROUP_LIST()}) exten => s,n,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) The GROUP_COUNT returns zero before the call to GROUP but also returns 0 after the call to GROUP. If I

To any that may be interested in the implementation of the conditional expression in the expression parser (ast_expr2*) in asterisk, I've filed the patch at: http://bugs.digium.com/view.php?id=4459 Right now, a comment has been added noting that the IF func provides this capability, and asks if both would really be necessary. It's a good question. I haven't been following the

Dear all, Anyone has experience using group and group_count to limit outgoing calls in AGI/CAGI? SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP EXEC Gotoif $[${GROUP_COUNT(OUTBOUND_GROUP@${CALLERIDNUM})} > 1]?BLOCK SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP But it doesn't work as it should. Tried in extensions.conf and it works. Any idea. Thanks Ray

Hello all, I was wondering what will be the "proper" way to manage BUSY state notification in presence once call-limit, incominglimit and all those settings are gone. I'm using GROUP_COUNT for call limiting in Asterisk 1.4.13 but I have no idea how to set up the settings needed for BUSY notification to work as I want it to. Basically, I want to disable call waiting (this is

Hello list, I'm using realtime queues and noticing that prompts are not played as expected. Database : announce = queue_youarenext = queue_youarenext queue_thereare = queue_thereare queue_callswaiting = queue_callswaiting queue_holdtime = queue_thankyou = queue_reporthold = announce_frequency= 10 announce_holdtime = announce-position = yes periodic_announce = periodic_announce_frequency =

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

From: Joerg Roedel <jroedel at suse.de> All drivers are converted to use the probe/release_device() call-backs, so the add_device/remove_device() pointers are unused and the code using them can be removed. Signed-off-by: Joerg Roedel <jroedel at suse.de> --- drivers/iommu/iommu.c | 149 ++++++++---------------------------------- include/linux/iommu.h | 4 -- 2 files changed, 29

the following is the code of the controller, under the dir components/test/: class Test::GroupsManController < ApplicationController uses_component_template_root def add_to_group @account = Account.find_by_nick(@params[:nick]) # render :text => "#{session[:account_id]} #{Group.find(session[:group_id]).owner_id}" if (session[:account_id] ==

From: Joerg Roedel <jroedel at suse.de> When a bus is initialized with iommu-ops, all devices on the bus are scanned and iommu-groups are allocated for them, and each groups will also get a default domain allocated. Until now this happened as soon as the group was created and the first device added to it. When other devices with different default domain requirements were added to the group

From: Joerg Roedel <jroedel at suse.de> This is needed to defer default_domain allocation for new IOMMU groups until all devices have been added to the group. Signed-off-by: Joerg Roedel <jroedel at suse.de> --- drivers/iommu/iommu.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/drivers/iommu/iommu.c b/drivers/iommu/iommu.c index

I have a simple cdr configured with the default tables, here is a row of a good cdr report calldate | clid | src | dst | dcontext | channel | ect ..... ect .... 2008-08-29 10:16:49 | "C. BOUTON" <40> | 40 | XXXXXXXXXXX | phonesystems | SIP/40-08776938 | ect ..... ect .... I have replaced the number by

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

Hi list, I have write down my code on which chanspy not working when I make a group with name of spy. Please help me where is the issue on that. a) caller will call this number to join konference and spy group exten => 43681111,1,Answer() exten => 43681111,n,NoOp(****${CHANNEL}****) exten => 43681111,n,Set(GROUP(${CHANNEL})=spy) exten => 43681111,n,Set(a=${GROUP_LIST(spy)}) exten

Hello list, using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first time... Having some troubles. This the dialplan (using a sub) : exten => s,n,Set(_custID=${custID}) exten => s,n,GROUP(${custID}) exten => s,n,NoOp(grouppcount = GROUP_COUNT(${custID})) exten => s,n,GoToIf($[ ${GROUP_COUNT(${custID})} > 2 ]?maxreached) The CLI shows : [Jun 5 16:06:26] --

Hello List, The list sorted my problem thus I shall contribute back ;-) PROBLEM: ======== I am posting this example, where I have a "Reunion" link of 30 channels. If i send all the traffic (proper + mobile) on the link, the less profitable proper traffic fills the link and leaves no channel for more profitable mobile traffic. Some kind of priority is needed to always leave space for

I have a system running Asterisk 11.2.1 that has had a couple calls between internal extensions get "stuck open". I didn't catch the verbose log for the first one, since I generally don't verbosely log to file, but the second one shows that the call that got stuck was dialed, but the caller hung up before the called device answered. This server is running a hotdesking

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm