similar to: Re: Has anyone tried using flash() in features.conf (applicationmap) - RESOLVED

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

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

hello, I use trixbox.I had define a feature code testfeature: [applicationmap] #include features_applicationmap_additional.conf testfeature => *3,callee,Macro,vote [featuremap] blindxfer => ## ; Blind Transfer disconnect => ** ; Disconnect Call automon => *1 ; One Touch Record atxfer => *2 ; Attended Xfer testfeature => *3 here is my macro-vote: [macro-vote] exten

I want to be able to send a prerecorded message to the person I am calling. I know that you can use the application map to do this. Just to test I enabled the testfeature example that is in the features.conf file. When I hit #9 during a call the other user does not hear the monkeys, they only hear a series of beeps. I have tried with different soundfiles and they all give the same problem.

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

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 ;

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

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

Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller => #9,callee,goto(521|1) show features - Dynamic Feature Default Current ---------------

Hi Guys!! I wrote a little patch for asterisk 1.2.5 and I will maintain it for future release unless somebody explains me how we can ask people at Digium to add it to the source tree... We are planning on using Asterisk as our main PBX for the office over the next few weeks. Our current setup uses TDM400 cards to bring our 8 lines into Asterisk, our Telco provides us an option for three

Hello, I have problems using dynamic features while an other person is doing call waiting in a call. I define a dynamic application mapping in features.conf as the following: testfeature => *9,caller,Playback,tt-monkeys I also set DYNAMIC_FEATURES => testfeature. The mapping is working well. But during a third person is calling I'm hearing just the call waiting tone and none of my

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, I found out that GoTo in applicationmap is not working. OK, LOCAL is working. but I expected that applicationmap is using the DIAL option tT. But it doesnt, it works without tT Option, so also callee can use internal functions if callee knows the code. Any workaround avaiable? best regards Thomas

Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

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'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

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 =>

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to