search for: conferment

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having

I have a site (for an academic conference) that I want to convert to RoR - both as a learning exercise and to ease further updates and development. There are a number of child pages for each conference (past and future), but these pages are different for each conference (not same child pages for each conference). The URLs are along the lines of: /conference/2000/keynotes

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of

First off, let me first say that this is not a one-way audio problem. Sometimes I can get 'her' to speak to me, other times I can't for a long time. I'm just using a very simple system to dump people into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf

Thought a different succinct subject line must drum up an answer or two... Also, this has been tested from two different carriers: We're getting an average of 2/10 call success rate. ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Thu, Aug 26, 2010 at 6:58 PM Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround()

[New Features] 1. Added focus and tab-order to all input fields 2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines

How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(<terminal>) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan dial

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

Hi there, I'm using asterisk meetme function like: exten => 9070,n,MeetMe(|dcM) and everything works pretty well. But I would like to add a review of the entered conference number before the user jumps into the conference. Somthing like: *:"Please enter the conference number followed by the hash key" (works) U: 123456# (works) *: "You are entering conference number

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? *

So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. extensions.conf: [meetme] exten => 2663,1,MeetMe(,De) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set a PIN. Hangup. Then users would dial

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

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

Hello, I was wondering if it is possible within Asterisk to be in many meetme conferences at the same time. This would be sort of broadcasting over all the conferences at once. Thanks, Denis

I have a did with 20 channels from didforsale. that we use to let local members call to listen to a conference several times a week without long distance charges. The upcoming call is getting more interest than usual and from places that are not local so we want to use a free conference service in addition to the local conference. How can I setup a conference on my asterisk box for the