similar to: Redirecting Calls and MeetMe Rooms

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx

Hi, Is it possible to create a meet me room on the go through dial plan? I am looking to use AMI Originate to drop a call into meetme room and once it's proved that party is joined, play him an announcement, grab few numbers from them, and then dial a second number and drop into the same meetme room. The reason to use this is to be able to know when the channels connected because both

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and they work fine when called from the legacy PBX. However, when there's a MeetMe room with a legacy caller and a SIP phone, the SIP phone can hear the legacy caller. But the legacy caller can't hear the SIP phone. However, "meetme show

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

I have encountered a frustrating problem with the meetme rooms and calls entering the system on the Digium analog cards. The typical scenario is: Callers on SIP phones, X-lite, Eyebeam, Cisco 7960, IAXy Callers entering the system from the PSTN via the digium Analog card (TDM400P) In the meetme room the SIP connections can all hear each other loud and clear. The PSTN people can hear

I have two asterisk machines. One of them has a Digium board (server A) and the other is simply using ztdummy (server B). Server A is running on Debian and Server B is running Gentoo. Server A is running Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running Asterisk 1.0.7. The problem I have is that when I try to transfer a call into a meetme room in server B, it simply hangs

I've googled and not quite found what I need, so... I have a conference room phone that I would like to make behave as follows: - when a call comes to that extension: answer the call put the call in a static MeetMe room with option 'w' ring the phone by SIP and when the phone picks up, put it in the same MeetMe room as the marked call. if subsequent calls come in, they are put

Hi, Matt and Asterisk gurus I encountered the same problem in my asterisk meetme. Whenever the 3rd person joins the meeting, it creates echo in the meeting, while 2 person meeting is fine. I am wondering if you can give me more hint on how to configure the mixer to have echo cancelled. We are using analog phones connected to asterisk TDM cards. Thanks a lot. Michael

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap

I want to have a "personal meetme conference", so when on a call I can transfer the other party to my personal conference with "#7". (I can then make other calls, and dump them into the conference using "#7" as well, then join myself by dialing "7"). Using: exten => 7,1,MeetMe(${CALLERIDNUM}|Mpd) this works as long as I originate the call. However,

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. cat meetme.conf [rooms] conf => 600 extensions.conf: [meetme] exten => 2663,1,MeetMe(,D) 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

I'd like to monitor key press in a meetme room. Is it possible when connecting one side of a local channel in the meetme room and the other side of the local channel to an agi with the command "wait for digit" ? Thanks Eric

Is there a way to have a Meetme room dial an extension? For example, is there a way to use the meetme as the channel in an originate command?

dear all, i facing one big issue in CDR , i have one user connected in meetme and then and after some time one user are joined meetme bu this user was originally dialed by , AMI via Originate and when remote user picked up call he joined meetme room everything is fine , but now 1 st user kicked 2 nd user , via using menu options or by typing "meetme kick 1234 2' user disconnected

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

Hi, This is pretty much a repost but I still havent been able to get MeetMe to work.. I am using a Dev Kit lite.. so that should satisfy the Zaptel requirement for MeetMe.. meetme.conf looks like this.. [rooms] conf => 7500 In extensions.conf I have an [extensions] context and within that same context I have the line.. exten => 7500,1,MeetMe(7500) When I dial 7500 I get the message

Has anyone a solution for me - with "Meetme(,Ms)" asterisk plays "conf-invalid" if a room not exist - with "Meetme(123,Ms)" asterisk plays not "conf-invalid" if the room not exist and asterisk hangup I am happy about any proposal. Thanks Daniel