similar to: features.conf : feature map ==> getting feature to work

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

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

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'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: 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)

I just installed the newly released Asterisk 1.4.7 and I cannot get music on hold. I am using the default settings with the wav files. Here is what I get on the cli from any sip phone: -- Executing [7001 at oficina:1] NoCDR("SIP/1120-084e6010", "") in new stack -- Executing [7001 at oficina:2] Answer("SIP/1120-084e6010", "") in new stack --

> I love this question, simply because it allows me to talk about one > of the neatest features I programmed into my system that barely > anyone knows exists. Plus it lines up pretty much exactly with what > you are trying to do. > > We have our gate control system tied into our Asterisk phone system > so it is possible to dial a code on the phone and open the entrance

Asterisk-server 1 : register => hostedasterisk:pass1 at domain2.biz [BOX-YOCAN] type=user ; incoming from YOCAN... auth=rsa inkeys=Key ; pub key of user secret=XXX ; pass2 of YOCAN context=from-BOX-YOCAN ; incoming from YOCAN trunk=yes transfer=no qualify=yes [BOX-YOCAN] type=peer ; outgoing to YOCAN... host=dynamic ; registers auth=rsa outkey=kEy ; private secret key

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'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

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 Kevin. Thank you very much for the hint! It worked very well! Your example ' exten => 1234,1,System(echo "This is a test" >> /var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - 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()

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'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

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

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 =>

Hi everybody, I would like to set asterisk up such that to use the call parking feature, instead of transferring a call to the extension set up in features.conf, you just dial a code (e.g. *3) and this then parks the call. 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

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

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