Multi Site Conference?

Thats an interesting idea.
If you are talking 1-8 users per site joining, then there might not be much point, but i suppose if you are going to have dozens of people per site joining the conference it might make sense.
Out of the box, won't work, but I can think of a couple of hacks.
Was this something you wanted the system to do automatically for every meeting?

I can have a coding idea in my head that might work. If someone has some ideas about it, post it. I'll code a hack this weekend and post it here.

FEATURE CODE!!

Heres how to do it (i think, in my head)...

Server 1: "Host" Server Virtual Conference 9001
-Inbound ROUTE with and CID Matching to 44444 (Any random, but understandable set of numbers) goes to conf 9001

Servers 2-Infinity
-Trunk using IAX to Server 1 with outbound Caller ID set to 44444
-Outbound Routes setup one called Conference and have it Only send out over the IAX and its pattern matching to something unique and short (ie: 88923)
- Feature code setup for 9001 to dial 88923, which will then get sent out over IAX with CID 44444 to server1, which will put user in conf.

You can work around with diff numbers and DID settings, but I dont see why this simple setup wouldnt work on all servers (As long as servers 2+ do not have an ext 9001)

Why couldn't someone at Trixbox 1 dial into Trixbox 2 and enter the conference through say and IVR? Then transfer that call to the conference room on Trixbox 1. This channel will inter-connect the conferences. Then your callers just enter the appropriate conference rooms. Call tear down might be a command line thing, or a custom script.

ahhh, i understand what you want to do now...

The transfer call to conference is prob your best method for bridging all the separate rooms together.

you may want to consider manually setting up such a 'tree' of conferences and evaluating the quality of the conference in such a scneario I could imagine the two tiered level of mixing that is going on in each conference 'leaf' and then again at the trunk of the conference might end up with problems. It can easily be evaluated by manually initiating the conferences and doing a test. This would be good background work to do before writing all the code to make it happen and then finding out the audio results are not good.

Welcome to "MultiMeet" !!

Here's some code you can try. Let me know if you find any issues, i'll try and fix it up.

Its BETA, but should work enough for you to try. I tested it with 4 PBX's and with about 25 users, seemed to work just fine.

WHAT IT DOES:
This allows for multiple sites, each running their own conference servers to be bridged automatically together as one large virtual meeting room. So instead of 3 centers with 25 users each all attempting to join a single server (so 50 outside SIP connections to a single server, plus the 25 local) this system keeps each of the 25 users on their own pbx, and only needs 1 single trunk between it and the 'master' server. So, for 3 PBXs, only 2 active SIP connections.)

OPERATION OUTLINE:
A user dials a ring group/ivr which points to the multimeet code.

It asks for the meeting number.
Person enters the number and the system attempts to join the meeting.
It simply detects the first caller into a meeting, then creates a 'star' type link to the other meeting servers it has in its configuration based on the Meeting number the first participant enters.

This creates a simple single trunk between the 'master' meeting room and the other pbxs.
(My code has the first PBX to have someone join the meeting to become the master)

HOW TO MAKE IT WORK:

Install the code (installation instructions further down)
Create a meeting room on each of your servers using FreePBX meeting GUI.
Create a ring group/IVR to point to CUSTOM, and in box to the right, point it to
multimeet-entry,s,1

Call that number, and the system prompts you for your meeting number
Put that in (currently defaults to 4 digits)
System will now join the meeting.
If it sees that this meeting has NO other participants, that triggers the system to make outbound calls.

INSTALLATION INSTRUCTIONS:

Open the /etc/asterisk/extensions_custom.conf file
copy /paste the code below the *************** in this post into that file
Save the changes and exit
type asterisk -rx reload
goto FreePBX
Create a ringgroup/ivr to point to CUSTOM and the context it will goto is multimeet-entry,s,1
Submit the changes
Click on the red bar
Create some meeting room numbers on your server (same meeting on each pbx)
(Repeat the above steps on each of your PBX's that you want to test it on)
Call the ringrroup/ivr on one of the PBXs and join the conference.

KNOWN OUTSTANDING ISSUES:

This code is BETA. Only tested marginally by me
This code won't disconnect the trunks between the meetings....yet.
I'll work on that code the next day or two.
Its easy enough for you to manually disconnect them, from the asterisk CLI
type soft hangup [Press TAB]
That will show you the active calls. You'll be able to see the calls that are linked to the meeting because it will have the same number as the meeting itself.

It doesn't handle meeting PIN numbers just yet, but I'll try and fix that (of course anyone ambitious out there can send me, or post additional code too.

***************
[multimeet-entry]
;Welcome to Multimeet Version .9 Beta
;This code is for instructional and training purposes only.
;This is the first module. Users come here first, they enter in their pin number
;At this time, pin numbers are assign ad-hoc. This will be corrected a little later on in the beta release.

exten => s,1,Wait(1)
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(5)
exten => s,4,Playback(enter-conf-pin-number&then-press-pound)
exten => s,5,NoOp(Requesting user to enter in their PIN number from their touch tone phone)
exten => s,6,Read(meetmepin||4|||5)
;this listens for the user to enter in their 4? digit pin number, with the # key, or 4 digits which ever is first
;they have 5 seconds to enter something, if that time is not met, the system will ask again.
;you can change this to reflect the meeting types you have.

exten => s,7,GotoIf($["${meetmepin}" = ""]?4)
exten => s,8,NoOp(the pin number entered is ${meetmepin})
;s,7 check to see if the user entered anything, if not, the system will loop him asking for a new number

exten => s,9,Goto(multimeet-2,s,1]

[multimeet-2]
;this module joins the user to the conference, and initates outbound calls to other conferences

exten => s,1,NoOp(Joining current caller to conference number ${meetmepin})
exten => s,2,MeetMeCount(${meetmepin}|count)
exten => s,3,NoOp()
;we now count to see the number of users in a meeting room to make sure that there are not 2 people in there already.
;if the system sees 2 or more in a meeting room already, we send them back to the beginning to select a new pin.

exten => s,4,Gotoif,$[${count} >=1]?100
;if the count is already 1(or more) then there is already a person in the room, or a MultiMeet link is established
;either situation would mean that making another link is not necessary at this time.

exten => s,5,system(echo "Channel: Local/${meetmepin}@from-internal/n" >> /etc/asterisk/${meetmepin}.call)
exten => s,6,system(echo "MaxRetries: 1" >> /etc/asterisk/${meetmepin}.call)
exten => s,7,system(echo "WaitTime:5" >> /etc/asterisk/${meetmepin}.call)
exten => s,8,system(echo "Context: from-internal" >> /etc/asterisk/${meetmepin}.call)
exten => s,9,system(echo "Extension: 89${meetmepin}" >> /etc/asterisk/${meetmepin}.call)
exten => s,10,system(echo "Priority: 1" >> /etc/asterisk/${meetmepin}.call)
exten => s,11,system(chown asterisk:asterisk ${meetmepin}.call)
exten => s,12,Wait(1)
exten => s,13,system(mv /etc/asterisk/${meetmepin}.call /var/spool/asterisk/outgoing)

;this code above creates a ".call file" that will be placed into /var/spool/asterisk/outgoing.
;This call file automatically generates a call within the asterisk PBX.
;The entry "Channel" defines the 'local' meeting the computer should join
;The entry "Extension" is the destination meeting it should join.
;This creates a 'trunk' between the two.

;On my two asterisk PBX's, I had created a meeting '6000' using FreePBX on each.
;In the above example, my 'far end' PBX is contacted with an outbound prefix of 89.
;So, when the person enters in the meeting they want to contact as 6000, the system will call 896000
;an outbound route in FreePBX simply of 89|. will send 6000 to the far end system.
;the far end system matches that to the local meeting and a link is made between the two systems.
;make sense? :)

;If you have multiple sites that you want to call, just repeat the code, modify the s,XX accordingly
;as well as the "EXTENSION" entry to reflect the outbound dial prefix that you need.

exten => s,n,Goto(s,100)
exten => s,100,MeetMe(${meetmepin}|Mds)

Based on some of the testing I did on this thing last night, Monster isn't even the right word. Colossal is more along the lines I think could be potentially achievable. The test I did last night was 200 simulated SIP users and 25 regular SIP users mixed with 5 PSTN lines across 5 boxes and it was fine. Worked like a charm and boxes weren't even breaking a sweat.

And the coding isn't that complex, I threw the previous posted code together in less than an hour. I just didn't bother with PINs or other things, but a little more time could easily do all the features such as waiting for Admins to join, minimizing on hold music, etc etc. Wouldn't be hard to do.

A little front end coding to spread callers around to different servers for some load balancing, I sincerely think it could be possible to achieve.

Lets pretend with Meetme that a box could handle 100 simultaneous meetme users (i will test what the limit is shortly, but i think it could do it) in either a single or several different meetings.....that means that you could have a single "HOST" box with 100 simultaneous SIP connections...if each one of those were hooked up to 100 boxes, each hosting 100 users each themselves....yowsers.

Lets say you were buying el-cheap $200 machines and doing everything via inbound SIP provider, for like $25,000 in hardware you could build a conference system that could, in theory, handle 10,000 simultaneous meeting place phone calls.

...Colossal...and inexpensive.

This is awesome.. thank you.

I could see a company that has implemented Trixboxes at each of their branches using this for a companywide conference call with ease.. Imaging putting 500 offices x 30000 users on a call vs 20000 users dialing into one server.

We currently have TB installed at 3 of our locations and I'm buying the hardware for the 4th. This is a great idea for saving bandwidth for conference calls.

I thought dmessano addressed this in the 5th post.

In terms of connecting TB boxes located in different geographic locations, the number of users is mostly irrelevant. The relevant parameter is how many TB's are connected to the main TB hosting the conference. It doesn't matter how many users are actually connected to each TB unit. In reality, I guess it could matter in terms of having to many local connections overloading your local 10/100/1000 network to your local TB unit.

Thanks, dickson for the code. I am installing it now and will test it amongst my boxes.

Just a thought on call teardown, and I am not a programmer, but when the user count gets to one in a given conference room, could you just go ahead and tear down the trunk call for that machine. If there is no human there, there is no need for that machine to remain on-line.

Thanks for taking time to write and debug this. It is an awesome tool.

I lost track of this thread for a while but I am still skeptical about the "infinite users" case. I follow the logic about the bandwidth saving but there is something "perpetual motion" in the logic and I am not sure where the disconnect is. I think the disconnect is that you will save the bandwidth as dickson's logic is correct, but as a result, the call quality will just degarde as the user count goes up.
I don't know the mixing algorithm at the server but there has to be data loss in the mixing algorithm otherwise mutiple people speaking from multiple sources down one channel cannot work. The more sources, the more the data loss required to keep it within a single channel. This would suggest that if you do in fact hook up a 100 users using four servers, the quality will be pretty poor and get worse as the number of users goes up, whether there is a single super server or 1000 average servers. Ultimately, at the server at the center of the star, all the end point channels are being mixed into single channel to be sent back to each users. So, yes, I think the bandwidth savings works but the actual call quality will make it unusable..... but I am of course open to a demo that disproves me :)

I can see where this should work fine with small numbers of speakers preaching to the multitudes or small multi site conferences. With just a handful of speakers allowed to be talking at once there will be no meaningful data loss. Maybe a good idea to extend the code to enforce keeping the number of active particpants (dynamically) limited.

Dickson,

I have created a manual way to tear down the trunks. You will notice I have shamelessly reused your code. Put this in extensions_custom.conf

************************************************

exten => 501,1,Wait(1)
exten => 501,2,DigitTimeout(5)
exten => 501,3,ResponseTimeout(5)
exten => 501,4,Playback(enter-conf-pin-number&then-press-pound)
exten => 501,5,NoOp(Requesting user to enter in their PIN number from their touch tone phone)
exten => 501,6,Read(meetmepin||4|||5)
;this listens for the user to enter in their 4? digit pin number, with the # key,
;or 4 digits which ever is first
;they have 5 seconds to enter something, if that time is not met, the system will ask again.
;you can change this to reflect the meeting types you have.

exten => 501,7,GotoIf($["${meetmepin}" = ""]?4)
exten => 501,8,NoOp(the pin number entered is ${meetmepin})
;s,7 check to see if the user entered anything, if not, the system will loop
;him asking for a new number

exten => 501,n,meetmeadmin(${meetmepin},K) ;The "K" logs off all users
exten => 501,n,hangup()

I see this thread is a few months old now... Has anyone deployed this into a prod environment, any feedback from users? What if any issues arose?

Sounds like a great solution to multi site conferencing and keeping it free/in house.

- Matt

Hi guys..
I was trying to work with this multi meetme setup.. It works great. accept for Meetme options like w, and A.
the leader arrived is not passed on through IAX trunk to the other server.

Any one has any clue ? or some one has already passed this hurdle

Nischal..

To make the code work in 2.6, replace the multimeet-entry part with this:

[multimeet-entry]
;Welcome to Multimeet Version .9 Beta
;This code is for instructional and training purposes only.
;This is the first module. Users come here first, they enter in their pin number
;At this time, pin numbers are assign ad-hoc. This will be corrected a little later on in the beta release.

exten => s,1,Wait(1)
exten => s,2,Set(TIMEOUT(digit)=5)
exten => s,3,Set(TIMEOUT(response)=5)
exten => s,4,Playback(enter-conf-pin-number&then-press-pound)
exten => s,5,NoOp(Requesting user to enter in their PIN number from their touch tone phone)
exten => s,6,Read(meetmepin||4|||5)
;this listens for the user to enter in their 4? digit pin number, with the # key, or 4 digits which ever is first
;they have 5 seconds to enter something, if that time is not met, the system will ask again.
;you can change this to reflect the meeting types you have.

exten => s,7,GotoIf($["${meetmepin}" = ""]?4)
exten => s,8,NoOp(the pin number entered is ${meetmepin})
;s,7 check to see if the user entered anything, if not, the system will loop him asking for a new number

exten => s,9,Goto(multimeet-2,s,1]

-----------------------------------------
Joshua Hagger
VANREE Software Consultancy
www.vanree.com