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Hi guys, I'm trying to connect Asterisk to the MySQL, but I can't execute it. It returns an error, as below: -- Executing [200 at teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200") in new stack [Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in

Hi - I'm trying to implement the applicationmap stuff in features.conf, and I can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom IP500s and Snom190s. My features.conf looks like this: [general] parkext => 700 parkpos => 701-720 context => parkedcalls parkingtime => 240 transferdigittimeout => 2 ;courtesytone = beep

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

Hi, Am running 1.4.18 at the moment, and am trying to implement inline blind transfer. I have : [featuremap] blindxfer => *6 ; Blind transfer in features.conf And in extensions .conf under [globals] : DYNAMIC_FEATURES=automon#blindxfr So what am I missing ?? Have read through http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Thanks,

Problem resolved. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back. extensions.conf [context] exten =>

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

HI This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext => 700 parkpos => 701-702 context => parkedcalls [featuremap] atxfer => *2 [applicationmap]

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com> wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no longer works. > > > > It only

I'm trying to actually use the example application map in features.conf: testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app

Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! -------------- next part -------------- An HTML

I am a newbie at RoR and am disappointed by the fact that I couldn''t install it in my Ubuntu 10.10 using an IDE. I tried NetBeans 6.9.1, Eclipse 3.6.2, Aptana Studio and its plugin for Eclipse and RubyMine. There was always a problem somewhere (rails, rubygems, rake, etc) preventing me from working with my project. I currently have Netbeans 6.9.1 bundled with java jdk 1.6.0. I''ve

Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/510@default-6b6c,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755]

Dear colleagues, Von Bertalanffy model is commonly adjust to data on fish length (TL) and age (AGE) TL= Linf*(1-exp(-K*(AGE-t0)). Linf, K and t0 are parameters of the model. One main goal of the growth study is the comparison of growth parameter estimates between sexes of the same species, or estimates from different populations. The realibility statistical tests normally applied are highly