similar to: Meetme define context

Sorry to re-post this but I'm sure it's something simple that someone has found before. To summarise: Dial plan answers the phone AGI script executes AGI debug in console show phonetics ABC - However no audio at all on the phone and this step is less than 1 second. Dial plan Busy Phone hangs up. Total time less than a second. This is such a simple AGI script do I need

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is.... # app_meetme.c:2593 find_conf: No Zap channel available for

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 updated Asterisk to 1.6.2.7 and now the user introduction in the meetme application is no longer working: [May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel type registered for 'DAHDI' -- Created MeetMe conference 1023 for conference '800' [May 27 09:26:51] WARNING[2407]: app_meetme.c:3640 find_conf: No DAHDI channel available for conference, user

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

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

Anyone seen this? If not I guess I'll have to post it as a bug. Extensions.conf has this: exten => 123,1,Meetme(|dMic|) I dial 123, and enter my conference number. Asterisk asks me to enter my name. At this point I hang up. If I type at the Asterisk console 'meetme list 12345' it shows that I am a participant in the conference evenhough I hung up. If I dial 123 again and this

Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme

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

Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear:

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>

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:

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, 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

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

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]:

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, 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