search for: applicationmap

...anging 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 => s,n,Set(DYNAMIC_FEATURES=zapflash) exten => s,n,Dial(SIP/xxxx,15,tw) features.conf [applicationmap] zapflash => *3,caller,flash,() <<<<<<<<<< needed a comma between flash an the () ----- Original Message ----- From: "John Voss" To: asterisk-users@lists.digium.com Subject: Has anyone tried using flash() in features.conf (applicationmap) Date: We...

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 transfe...

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

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: [m...

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

...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 callee to extension 123. I'm under the impression this must be done with dynamic feature (applicationmap) and ChannelRedirect() because of the SCCP protocol. The result is that my test code, which is far from finished, only works occasionally. What happens when it fails is that the callee is hung up instead of put into the dynamic conference. There doesn't seem to be a reason for it to fail when...

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 fellow asterisk users, I'm trying to use feature application defined in application map. it's defined as follows: lbxvml => 1,self/caller,Macro,Jump2Voicemail It's working properly when called party answers the call, but I'd like to have feature usable while call is still ringing in queue but this just does not work.. Is this a bug or feature? Is there a way to have

...d maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon => *1 I tried also to use the [applicationmap]] but it doesn't seem to work. Pressing #1 do nothing. Here my dialplan: 9999 => { Set(__DYNAMIC_FEATURE=pauseMonitor); MixMonitor(test); Dial(SIP/1000 at srv01,30,TtX); } [applicationmap] pauseMonitor => #1,self/both,stopMixMonitor...

...: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb) exten => _X.,n,Set(__DYNAMIC_FEATURES=in-pauseMonitor#in-resumeMonitor) exten => _X.,n,Dial(SIP/myphone,300,tTo) My [applicationmap] in features.conf is setup as follows: in-pauseMonitor => *7,self/callee,Macro,pause-record in-resumeMonitor => *9,self/callee,Macro,resume-record I also have the following contexts setup in extensions.conf: [macro-pause-record] exten => s,1,Playback(sounds/recPaused) ext...

...The main reason for this is that a number of the phones I use have transfer buttons that I can't reprogram to use Asterisk's own transfer functions, therefore you don't get the announcement of which extension they've been parked onto... I assumed the way to do this would be the applicationmap in features.conf, so I tried various variations on this: parkcall => *3,caller,ParkAndAnnounce,pbx-transfer:PARKED|60|Local/${EXTEN}@internal|internal,${EXTEN},1 The most obvious problem I have had is that ${EXTEN} isn't decoded, I couldn't find much documentation on the application...

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 scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100626/f006be0d/attachment.htm

...tion. 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 work. Steps. 1. added the example macro to the dialplan in extensions.conf 2. added the line MixMonApp => *1,self/both,Macro,mixmon to the features.conf file under [applicationmap] 3. sip reload / dialplan reload / reload res_features 4....

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 working....

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation b...

Hi, Does anyone know how to implement the feature in asterisk calling card when a user has dialed the access number and during the IVR or any time during the call, he can press ## or ** to end the current call and dial a new destination number? Please help and give me a step by step help. Thanks. Rgrds-------------Abid -------------- next part -------------- An HTML attachment was

...o be touched. When the button be touched, during an ongoing call, the software (SIP client) will sends a request to Asterisk executes the "gate => 9,self/callee,System,<insert command here>" , for example. So, it will works like the user pressing number 9. I will take a look at applicationmap in features.conf to understand what exactly can be done. But, let me ask you: This idea seems to be good to run during ongoing calls. What about moments when there is no ongoing call? That is, can Asterisk execute a dial plan (maybe by means of some kind of SIP request received from the SIP client...

Hi, For thanks to everyone that answered the "dial from pph". On an other subject, how would I go about playing a wav file while talking to someone over a Zap channel ? Let me explain. I am on line with someone. I want him to hear a WAV (or mp3) sound file. I punch a key on my phone keyboard and he hears the sound file and after we can continu talking. Any hints

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 --------------- ------- ------- dumpcaller no def #9 Result.... -- Feature Found: dumpcaller exten: dumpcaller Dec 1 20:08:02 WARNING[18659]: res_features.c:958 feature...

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil