similar to: Help with playing a recorded message in a conference.

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

Not sure if this is the right place to ask, but what do we need to do to get this patch merged? How can I help? I'm no dev, but I use LDAP with Asterisk and I might be of some help. Thanks guys.

Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial

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, 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 "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'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, 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 all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

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.

Hello, in advance: You can also answer me in German. I know Icecast over 10 years very well. My problem is this and I am trying to describe it very short: If my Icecast stream/mount is offline, a recorded MP3 file should be played via the Icecast mount. The mount is configured as follows: <mount>     <mount-name>/LiveStream1.mp3</mount-name>    

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

Hi, For question 3, I think there is ezstream (https://icecast.org/ezstream/) which can to an endless loop like you wish. All files should (actually must) be encoded in the format of your stream. Also I would setup the mpuntpoints in a row: - live mp with fallback to ezstream mp - ezstream mp with fallback to static file So if ezstream fails you deliver the static file. Most notable in your

I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

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:

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

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, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All