similar to: Group count not being preserved when transferring a call into a conference

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

Hi, I am having some trouble with parked calls timing out. In features.conf: [general] parkext => 800 ; What extension to dial to park parkpos => 801-820 ; What extensions to park calls on context=parkedcalls parkingtime=120 After the Park timesout it calls the phone that the call was parked from. If the phone is BUSY the call just get dropped. (Call waiting

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

Hello, this is a test to add a channel to multiple GROUPs. this is my dialplan : exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)}) exten => s,n,Set(GROUP(40)=40) exten => s,n,NoOp(This channel is member of : ${GROUP_LIST()}) exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)}) exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40L)}) exten => s,n,Set(GROUP(40)=40L) exten =>

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

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

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

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, 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

Hi All, I got this error: [Jan 18 09:56:58] WARNING[9617]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: 1 > ^ exten => s,3,GotoIf($[${GROUP_COUNT(${ARG1})} > ${LINELIMIT}]?101) exten => s,3,GotoIf($[${GROUP_COUNT(${TRUNKGROUP})} >

Hello! Names of groups should be stored in backend memory pool. --- acl-backend.c.old 2008-10-02 11:43:05.000000000 +0400 +++ acl-backend.c 2008-10-02 11:43:48.000000000 +0400 @@ -62,7 +62,7 @@ backend->groups = p_new(backend->pool, const char *, group_count); for (i = 0; i < group_count; i++) -

I want to limit the number of simultaneous incoming calls that my IAX DID can accept to, say, 2. The IAX DID provider sets no limit. The code below does work, but when the limit is in effect, new callers hear a "call cannot be completed as dialed.." message instead of a busy signal. Maybe this is an issue with the provider, but I do not like this and want callers to hear a busy signal.

Is it possible to increment a call group (Max_Calls) from one server to the next server? I'm trying to set a maximum call limit between two facilities (2 Asterisk servers) using Set Group. I want the limit to be set at 4. The problem with the code that I have in place, is it allows for 4 on each server, hence the possibility of maxium of 8 concurrent calls. I need to be able to, when 1

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

Hello users! Recently I read that AgentCallbackLogin is going to be deprecated soon. Wanting to set up a few callback type queues, I set them up as suggested in queues-with-callback-members.txt. I was able to set the queues up completely this way, however, I'm trying to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login status. FOP monitors their status if I call