search for: featuremap

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...

Hello all. Dialing with tT options and function Read (to prompt number) has a trouble for me. Can I temporarily features off during Read? features.conf: [featuremap] blindxfer => ## ; Blind transfer (default is #) atxfer => ** ; Attended transfer I try: exten => s,n,Set(LOCAL(tmp_atxfer)=${FEATUREMAP(atxfer)}) exten => s,n,Set(LOCAL(tmp_blindxfer)=${FEATUREMAP(blindxfer)}) exten => s,n,Set(FEATUREMAP(atxfer)=...

Hi there i just can't find an answer on the featuremap config i want all phones to use the same method for transfering a call on all phones but i just can't get the atxfer or other functions to work on my grandsteam and sipura spa 2000 it's confusing for users with different phones to transfer a call i know you can use the transfer button but...

Hi, I am new to asterisk configuration. I want to get called number in features.conf. I am defining a feature in features.conf and that feature got executed on pressing a particular DTMF key sequence. As I want to execute my own application on pressing that key which will use called number. testfeature => 3,peer,AGI,StoreNumber|<CalledNumber> Here I want to use called number in place

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this: Dial("SIP/14165551212@<MyProvider>,30,TgH,"") This Dial command is buried in a context that is called using the Gosub command. When I press ** the Dial command exits and processing continues as expected given the &...

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and 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 no...

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here http://www.voip-info.org/tiki-index.php?page=Asterisk+Down...

...uck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup() [disabletransfer] exten => s,1,Set(FEATUREMAP(atxfer)=) exten => s,n,Set(FEATUREMAP(blindxfer)=) exten => s,n,Set(FEATURE(inherit)=yes exten => s,n,Noop(${FEATUREMAP(atxfer)}) exten => s,n,Noop(${FEATUREMAP(blindxfer)}) exten => s,n,Noop(${FEATURE(inherit)}) exten => s,n,Playback(beep) exten => s,n,Return() I can see that...

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features lik...

Hi, Is there any way to run a command (AGI script, whatever else) at that moment when the call that was in the queue is being bridged to a specific agent? An examples of what I would want to do with this is, for example, have Asterisk ask the caller for his 4 digit customer ID before being put in the Queue. Once I know who the caller is being connected to (which agent) I'd run a

Hi, I'm using the new AMP which provides a call recording. The options of recording call Always and Never are well working. But how to use the On-Demand option ? Should I press a pad ? Is this configured in the featuremap of features.conf ? Why my modifications in that features.conf have no effects ? Please advice me. Alexis.

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

...and I have the same settings on a 1.4 server, and the music plays correctly when using the same phone. When using the * 2 to transfer the connection or a softfone, the music plays correctly on this server with Asterisk 1.6. What the detail is missing in my configuration? My Configuration [featuremap] blindxfer=## atxfer=*2 automon=*1 disconnect=** I made a DEBUG to use the channel when the two key TRANSFER Server 1.4 and 1.6. Command: core set debug channel SIP/2720-b7d28d70 DEBUG no 1.4: << [ TYPE: Control (4) SUBCLASS: Unknown control '16' (16) ] [SIP...

...also tested queues and they work without a problem, too. Next up is my extensions configuration, of which the interesting section now looks like this: [default] include => general ; longshot, added out of desparation include => parkedcalls ; longshot, added out of desparation include => featuremap ; longshot, added out of desparation exten => 800,1,Answer exten => 800,2,Dial(SIP/frank,20,tT) exten => 800,3 Hangup exten => 802,1,Answer exten => 802,2,Dial(SIP/test,20,tT) exten => 802,3 Hangup Notice the inclusion of several contexts that should or would have to be define...

...the moment I was not successful. I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer => #1 atxfer => *2 verifying previously that the extension uses the arguments "tT" with the Dial application and to include the context "featuremap" in the context in which I have defined the extensions (internal). The telephone of the end with which the conversation is staying listens the tones to try doing the transfer, but Asterisk does not give the dial tone after *2 / #1 or the recall key. I copy my configuration files after to hav...

...ck -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') ....... extensions.conf: [globals] ....... GOTO_ON_BLINDXFR="internal,266,1" features.conf: [featuremap] blindxfer => #1 But: -- Executing [s at DialOut:14] Dial("DAHDI/1-1", "motif/xxxx/+1234567890a at voice.google.com,,rTt") in new stack -- Called motif/xxxx/+1234567890a at voice.google.com -- Motif/+1234567890a at voice.google.com-688c is proceeding passing it...

...sfer ;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] ; there are different from the default in features.conf.SAMPLE, ; and faster to type when apropriate, OMHO blindxfer => #7 ; Blind transfer disconnect => *0 ; Disconnect automon => *1 ; One Touch Record atxfer => *7 ; Atte...

Hello, while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers a "featuremap" to assign a key to blindxfer and atxfer and they work fine if the call is still in the same starting context, but if the call has moved in another context, then the new call will be started from such context with unpredictable results. Do you have any idea to make all transfers to be applied...

Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer.... also can

...ntil I realized the "transfer" only works on xPRO, which isn't available for linux. So I cant rely on SIP to handle this, I set up my extensions.conf have transfers, ie: [sip-exten] exten => 1001,1,Dial(SIP/1001,20,Trt) exten => 1001,2,Hangup And features.conf is : [featuremap] blindxfer => *1 ; Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record atxfer => *2 ; Attended transfer OK each analog phone has three way calling on it ... can I set up a "flash" com...