similar to: Scalability and stability matters

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

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

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

I have a very old server that is used only for conferences on Meetme. To manage the conference rooms we use Web Meetme. Now it is time to upgrade everything but since Meetme is no longer available I need to find a replacement GUI to manage the conference rooms. Anyone know a solution that works with Confbridge? I found "Asterisk Confbridge Manager" from a russian company but it

Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd But if I make the

Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created. The rest of the week,

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

Hi all, How scalable is asterisk ? I am considering using asterisk as a VoIP platform/gateway between Internet and PSTN (switches) to offer services to home customers. What goes along with it is eventually a lot of users - upto thousands probably. Is load balancing possible with multiple asterisk boxes ? Does anyone have any sort of info/experience with such projects ? How would asterisk cope

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 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

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 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'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, 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

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me