similar to: applicationmap and ChannelRedirect

Hello, I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with ChannelRedirect. I have a caller connected to an agent. The agent may request additional help by consulting another department. I can't use manual process with blind or directed transfer as the agent have many different numbers to dial. The message with the proper dial number is coming from the host. I got

Hi, i'm trying to implement the following scenario: - A user calls number 700 - Asterisk then dials to extensions 100, 200, 300, 400 and 500 - And then bridges all calls to a conference room I tried to use MeetMe and ChannelRedirect, but seems that after channel redirect nothing more is executed. So, this seem to work for the caller and first called, but the others

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use

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

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.

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

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'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Guys, I am using Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux on 2013-01-18 19:52:25 UTC How can I set variable in one context and then Redirect a channel to another context and use variable there? The code below doesn't work, so I've got empty VAR1 in context_2 [context_1] exten => s,1,SET(__VAR1=VALUE1) exten =>

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

Hi, My problem is as follows. I registered an xlite client and dialed 1500 extension. In the extensions.conf i set as follows. exten=>1500,1,AGI(localhost// hello.agi. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work.

hi, I hope someone can help me:-) we’ve got a freepbx server. there are 2 special extensions (2001, 2002). if someone calls this extensions (or a call is forwarded to these extensions) and these extension hangup (not the caller party), then we’d like to put the calls back into a queue (1000) and wouldn’t like to hangup. I read your description about hangup hooks:

Is there a way to make ConfBridge hang up on the final participant in a conference (obviously after some sort of initial grace period)? Background - I have just moved all of the phones in my house to separate extensions. As a replacement for the POTS-style shared line, I have implemented a "barge in" feature; any internal extension is able to join the call of any other internal

I have a Bridged call with 2 parties. I want to redirect one party to a conference room and the other party to an outside number. I tried doing that with a dialplan. I used ChannelRedirect in the dialplan and redirected the first channel to the conference room. however, the second channel disconnects. Reading thru the mailing list i understand that this expected. However, I don't understand