similar to: Teleconferencing using Zapta cards.

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

Just came by this link So I'm posting to keep the community informed. I don't use or endorse this product. I'm just letting people know about it. http://www.indosoft.ca/audioconferencesystem.htm Audio Conferencing System & Teleconferencing Solution that connect seamlessly over TDM and IP networks. This audio conference system include a comprehensive set of features and easily

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

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

I'm trying to use Asterisk (v1.2.11) make a callback that dials both legs of the call into a Meetme() room together, but I keep getting "conf-invalid" messages. I created a callfile (/var/spool/asterisk/outgoing/out.call) that specifies a Local channel (extension) which contains a Dial() command to the "dialer", and an extension which contains a Dial() command to the

Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created. The rest of the week,

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Hello , I am new to Asterisk. Trying to use Meetme for Audio Conferencing. Got Zaptel card etc. and i could see app_meetme.so nicely loaded. Now : 1. how to start a conference ? 2. how to add a user ? 3. How can a user join a conference ? After looking at certain links on Net I tried to

Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

I was able to get it to work with 2 extensions. One for the "host" and one for the "participants" Not the best way to set it up but it works. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, December 13, 2006 8:06 AM To: Asterisk Users

A couple comments. I'm not a programmer, my C is passable, but my web development would have to grow by leaps and bounds to be considered poor. I pulled the Meetme2 from here: http://www.areski.net/asterisk-meetme/about.php?s=0 The app needs a minor tweak to compile against 1.0.5. I stumbled down a false path or two, so the diff shows some lines being deleted and re-added that are

Hi all, How scalable is asterisk ? I am considering using asterisk as a VoIP platform/gateway between Internet and PSTN (switches) to offer services to home customers. What goes along with it is eventually a lot of users - upto thousands probably. Is load balancing possible with multiple asterisk boxes ? Does anyone have any sort of info/experience with such projects ? How would asterisk cope

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

Greetings, I have a somewhat unique (I think) configuration that I am testing involving MeetMe conferencing and have encountered a problem that I'm not quite sure how to solve. Here is a brief description of my setup for the background. I wanted to offer the ability for users to mute and unmute themselves while in a conference. If they enter a conference as monitor only, they are

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

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? *

I've been asked by my company to setup a conferencing system to support up to 400 people on a conference calls, where all users will be dialling in frpm the PSTN. I am exploring using Asterisk meetme to do this. I have two questions in relation to this:- For Meetme conferences is it better to have all participants to dial in via SIP provider terminating to Asterisk via SIP/IAX, or use

Hello, all. I've solved my own problem but will post it here in case someone else has the same misunderstanding in the future. We thought we had set up our meetme so that regular users entered the conference without a pin but could not speak to each other until the moderator arrived. We enforced pin entry on the moderator . . . or at least so we thought. If the moderator waited long enough

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after