similar to: MeetMe application in Asterisk V1.07

I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten => 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the

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

Hi All, I'm still having trouble trying to figure out if it is possible to define (in the dial plan) a context for meetme? I'm using 1.4.4 with dialplan logic of: exten => 123,1,Meetme(,Msa,) This defaults to conferences defined within the rooms context of meetme.conf Is it possible to specify another context as with voicemail? Or can any one think of another

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

Hello, I have MeetMe rooms generated dynamically and it always have two people inside that are entered by dialplan. I wish to make in some way a timeout mechanism that after X amount of time, it will disconnect the users and kick them out of the conference. How can I do such thing ? Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL:

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

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'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference "room", ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

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

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:

Hi guys, Having a bit of a problem trying to get conference bridges working. In my meetme.conf file I have the following line [rooms] conf => 6000 In my extensions.conf file I have: exten => 1000,1,MeetMe,6000 My problem is that when I dial into extension 1000 it is telling me "this is not a valid conference number". Can anybody telling me what I'm doing wrong here?

Hi all, I'm doing some tests with the Meetme application, unfortunately, I can't make it works. If the application list "Meetme" doesn't exists. meetme.conf: [rooms] conf => 8600 extensions.conf [default] exten => 8600,1,Meetme,8600 In the console writes: WARNING[1200825920]: File pbx.c, Line 1051 (pbx_extension_helper): No applications 'Meetme' for

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

Hi, Matt: What?s your server specifications that did you use? Best Regards, Cleviton. -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Matt Florell Enviada em: sexta-feira, 23 de junho de 2006 11:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Meetme max users

Is it possible to record a meetme conference? What channel would you monitor, is there a main channel that all audio goes too? If so, is it possible to use the ast_monitor (iirc) to record that channel? Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 =>

New server, running 1.2.9.1 with a Sangoma 4 port T1 card, wanpipe rev 2.8.4. I get this message after using a MeetMe room a few times: Jun 27 18:37:23 WARNING[1054] chan_zap.c: Unable to open '/dev/zap/pseudo': No such file or directory Jun 27 18:37:23 ERROR[1054] chan_zap.c: Unable to dup channel: No such file or directory Jun 27 18:37:23 WARNING[1054] app_meetme.c: Unable to open