similar to: meetme don't play conf-invalid if room does not exist

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

Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the room. It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ? can anyone help me and maybe someone has also the problem as i and have an solution. I use: asterisk-1.6.2.7

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

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

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

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 can't seem to locate any documentation on what this does. I tested it out with a simple static conference room: exten => conference,1,MeetMe(,1aMqw) and a static room defined in meetme.conf: conf => 123456,22,1 Users can get in with either of the pins, but I don't see that it does anything - I can't access the admin menu, nor does it set the user as marked to open up the

Hi @ all, what is the best way to to use features like MeetmeCount without leaving the conference. I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the caller leave the Conference :( Is it possible to press a key, without this obstacle? Thanx for your answers Daniel Knoll

Hello. I have a strange problem with the MeetMe application. Configured is a misdn msn to go into a preconfigured MeetMe room. exten => 12,1,MeetMe(1234,pIM) The first caller gets the prompt to enter the pin and then gets connected to the MeetMe room. The second caller gets also the prompt but after entering the right key he hears a dialtone followed by the message: The number you have

Hi guys, i'm trying to use the "featuremap" of features.conf inside the app meetme, but it's no working. like: _5XXX => { Set(DYNAMIC_FEATURES=toca_macaco); MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF Hangup(); }; in features.conf: toca_macaco => 123, peer, Playback,tt-monkeys But, if, inside the room, I press *123* the sound file

Hi, I'm new to asterisk and have a little trouble in developing my first more complex dialplan. The basic task is a click to call solution: - call one number via sip, play some announcements, do cdr etc. and put the callee into an conference room with music on hold - call a second number via sip, play some announcements, do cdr etc. put the callee into the same conference - have a nice chat

Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002", "CHANNEL(language)=fr") in new stack -- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002", "") in new stack -- Executing

Hey Guy! I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten =>

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

Hello everybody, using the manager api (via asterisk-java) I originate a call with application MeetMe to some extension (IAX). The agent joins the meetMe room on answering that incoming call. So far so good. Now I'd like to redirect that agent from the meetMe room to another meetMe room *only by using the manager api*. Is that idea possible to realize? Or has the agent to be involved?

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

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It

Hi Group, i have a MeetMe Question. I use "MeetMe(,Ms)" in the Dialplan and if a Conference Room does't exist Asterisk play (conf-invalid.slin) If i use "MeetMe(${room},Ms)" (value from DTMF Read) and the Conference Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup the Call. there is a solution for the kind my problem? Thanx and

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