similar to: Calls getting "stuck open"

I run a hotdesking system based on the example from Asterisk: The Definitive Guide. Calls come into the [hotdesk] context, which verifies the phone has a logged in user and sends the call to users,${EXTEN},1 if there is a user logged in. The [users] context then includes several other contexts for internal/external call handling, as follows: [users] include => internal include =>

Has anyone managed to do this? I thought about using sessions but they dont want to work for me... Here''s what I''m doing as a little test: class NotLoggedIn < Exception end class ProjectsController < ApplicationController wsdl_service_name ''Projects'' web_service_api ProjectsApi def Login @session[:loggedIn] = true end def GetProjects

Good evening! I''m currently trying to submit some data and a file upload through Ajax. I''m using rails & prototype, and somewhere there''s a disconnect. I''m reasonably convinced the file upload isnt being sent at all, or if it is, Rails has absolutely no idea. The form itself heads with: <form onsubmit="new

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

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

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

Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use

On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna <jlamanna at gmail.com> wrote: > > On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> > wrote: > >> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> >> wrote: >> > Also, how big does the cache in frame.c grow to? >> > I've recompiled with

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 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 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 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! 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++) -

Hello, I followed the steps by Russell* http://www.venturevoip.com/news.php?rssid=1980* and I got it working for publish_event only. As soon as I add subscribe_event, Asterisk doesn't start and I just get the following message: *Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node ffffffff* I have no idea what's wrong. There is not very much information about this issue. Where

You can use a custom device state to do it. [dnd] ;DND Toggle exten => *363,1,Answer() same => n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})}) same => n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1) ;DND On exten => *78,1,NoOP(Turning DND On) same => n,Set(DEVICE_STATE(Custom:DND${CHANNEL(peername)})=BUSY) same =>

I'd recommend using DEVICE_STATE On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not 'NOT_INUSE' then dial it, Otherwise dial SIP/102 exten => 101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40)) same => n,Dial(SIP/102,40,t) same => n,Hangup() On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

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