similar to: Conference Call Query?

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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.

I have a group of local users. They want to participate on a conference call to a PSTN line in USA. To connect they need to enter a code there as well. I want to use a local conference room, where LAN users and local users can call in. The conference room should be connected to the conference abroad. features.conf has: [featuremap] blindxfer => #1 ; Blind transfer ;disconnect

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

I'm just figuring out conferencing. I have a super-simple setup with one room: exten => 600,1,Answer exten => 600,2,ConfBridge(1234,c|M|s) exten => 600,3,hangup If two people want to take their (bridged) call to the conference room, the local user has to do a transfer (to 600), and then dial 600 themselves. Is there an easier way to transfer both ends of a bridged call to the

Hi, I am organizing a Virtualization track at LPC 2010 (Linux Plumbers Conference 2010). Please see the official call for tracks below. LPC is particular well suited for work in progress and subjects that needs discussion and collaboration between communities, for example between KVM/QEMU and the Linux Kernel community. We expect strong participation of kernel developers at LPC, and likely also

Hi, I am organizing a Virtualization track at LPC 2010 (Linux Plumbers Conference 2010). Please see the official call for tracks below. LPC is particular well suited for work in progress and subjects that needs discussion and collaboration between communities, for example between KVM/QEMU and the Linux Kernel community. We expect strong participation of kernel developers at LPC, and likely also

Greetings, I have created simple conferencing solution before using meetme application, but this times its a little tricky. My client needs a functionality to call multiple extensions to join a conference room. Extensions will ring like in a ring group, and on pick up, user will be either automatically added to the conference room, or maybe I'll program them to enter 9 to accept and 8 to

Hi all, I'm the main organizer of the Open Help Conference & Sprints. Open Help is taking place September 26-30 this year, its fifth year. Open Help features a combination of talks and open discussion on topics related to open source documentation. The conference portion is September 26-27. After the conference, we host three days of multiple team doc sprints, September 28-30. We have

I have successfully set up a conference room on my asterisk server, I have been trying to make the 'M' for music on hold option work (when the first person enters the room they are told they are the first and then they are supposed to hear music on hold) but it didn't matter which way I wrote it this feature wouldn't work. Basically it wouldn't allow the conference to be

I'm having some problems with Meetme2 and MOH. Here's what I've done, not sure if it's correct. I added the following line to extensions_additional.conf exten => 3000,1,MeetMe2 And I added conference rooms to Meetme_additional.conf conf => 8005 conf => 8006 conf => 8007 If I dial directly into my conference room by dialing 8005, I get "you are currently the

Hi, In Asterisk, what happens to the files when both legs of the call hangs up? Is there a way to create a conference room on the flight? i.e. without pre-defining the conference ID in meetme.conf. Thnx much. -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 2872 bytes Desc: not available Url :

Dear List I have implemented application for my client which include calling 2 end points and bridging them in conference. now end point A is local sip extension. and endpoint B is cellphone number which is called via sip trunk. I does this by AMI with originate action. now issue is when i try to place call if user is not reachable or switched off then i dont get to know instead i get only one

When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number

Hi: Can I set 1 extension(i.e.6000) in extensions.conf file for several room for conference call service ? Or for every room I should set 1 special extension. Regards. --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radio station. I assigned "mono out" in XP Equalizer as the mic so now I have the softphone streaming audio. I then create a conference room and dial that room from the softphone. Now anyone who joins the conference room hears the streaming audio. How can I configure Asterisk so that

Hi, I'm trying to configure a system with dynamic's conference-room creation. Anybody knows whats the limit of conference-room supported by asterisk?? and the rang of this conference-room (number)?? Thks a lot.