How To Set Up A Conference Call Bridge

Knowledge Is Power

An Like shooting fish in a barrel To Build
Free Conference Bridge

A conference bridge allows multiple users to all dial the same phone number and all be connected together as if they were in a virtual conference room. A single conference bridge can have multiple "rooms" with each room having information technology's own telephone number. Typically, teleconference bridge functionality built into corporate telephone systems from manufacturers such as Nortel and Avaya were extremely expensive. They realize that conference calls tin can save companies a lot of money in time and travel so they accuse a premium for the power to host such calls. The most mutual culling to expensive PBX manufacturer options are hosted teleconference services with either monthly subscription charges or per-minute/per-user charges which can also get expensive.

Asterisk, the open source software PBX, has long offered MeetMe conference span functionality. The problem is getting it to interface with your corporate phone system. 1 common option is to interface an Asterisk system to a PBX using a SIP trunk. The choice we'll utilise on this page is to fix up an H.323 connexion between Asterisk and an Avaya IP Office PBX. However, it should work for any PBX that supports H.323 connections. We chose to utilize an H.323 connectedness considering it is a more robust protocol (should you wish to aggrandize beyond elementary audio teleconferencing to teleconferencing with video and white board capabilities at some betoken) and, on our phone system, it didn't require any additional licenses (we would accept needed to purchase SIP licenses to use a SIP trunk).

At that place is an all-in-ane version of Asterisk called AsteriskNow that includes not only the Asterisk PBX software only the CentOS Linux operating system and a Spider web-based GUI management interface for Asterisk called FreePBX. Y'all but boot off of an AsteriskNow DVD and everything you lot need for a fully-functional PBX system gets installed and mostly configured for you.

Here'southward all yous'll need to do to get a conference span working with an H.323-capable PBX:

1. Download the gratis AsteriskNOW DVD ISO image and burn it to a DVD.
2. Install AsteriskNOW on PC or server.
3. Use the FreePBX GUI management tool to configure a conference room.
4. On the Asterisk server manually create an H.323 configuration file.
5. On your PBX create an H.323 torso.
6. On your PBX create a short code (road).

Download AsteriskNOW

You tin can download the latest AsteriskNOW DVD ISO image from here. Naturally if you're going to be using an older PC y'all'll want to download the 32-bit version.

www.asterisk.org/downloads/asterisknow

Install AsteriskNOW

NOTE that installing AsteriskNOW will wipe out whatever is currently on the hard-drive.
The DVD you created using the paradigm you lot downloaded is called a "Asterisk distribution" because it includes everything, the Linux operating system (CentOS), the Asterisk server application, the Apache Web server application to serve upwards the FreePBX Web pages, etc. etc. etc. Installing the software off of this one DVD will give you a complete Asterisk PBX server.

While fifty-fifty an older PC will work just fine to gear up a examination or pocket-size conference span, if you plan on setting upward a production briefing span that volition see moderate to heavy use be sure to accept into business relationship the maximum number of simultaneous callers you lot'll take on the system. The number of rooms or the number of callers in a room doesn't affair equally much as the number of all callers put together. If you're going to have fifteen or more people simultaneously dialed into the organisation you'll want to install the software on some pretty respectable server hardware with a fast CPU and a lot of retention.

Screen shot courtesy of the AsteriskNOW Installation Folio
Kicking off the DVD and the AsteriskNOW installation menu will appear. With Full Install under the newest version already highlighted press Enter.

Screen shot courtesy of the AsteriskNOW Installation Page
The next screen requiring input is the "Configure TCP/IP" screen. If you're not accustomed to navigating text-based screens you'll want to printing the tab key to move the small highlight to the asterisk in front of "Enable IPv6 support" and printing the infinite bar to de-select information technology. Then printing the tab primal once again to highlight OK and press Enter.

Screen shot courtesy of the AsteriskNOW Installation Folio
Likewise on the "Time Zone Pick" screen, press the tab key to highlight the listing of time zones and utilise the upward and down arrow keys to discover yours. You typically won't find "Eastern" or "Cardinal" so you take to select a metropolis (such equally Chicago for Usa-Cardinal). Once you have your time zone city selected tab to OK and press Enter.

Screen shot courtesy of the AsteriskNOW Installation Page
Side by side you are prompted for a password. You will exist prompted for ii passwords during the installation and setup. This is the starting time one and it is for the Linux root account. On a Linux system root is the super-user account, like Ambassador is on a Windows machine. You lot demand this password to log into the system itself so don't forget what you enter hither.

The 2nd password y'all'll be asked for is for the FreePBX GUI management application. You'll not simply exist prompted for a countersign merely for a user proper noun besides. In that location's no reason you can't use "root" for that user proper noun and the same countersign just to proceed things unproblematic.

At this point your piece of work is done for the software installation function of the setup. All of the Linux, Asterisk, and FreePBX awarding packages will be installed automatically. The installation may seem like it hangs at diverse points, especially if you're using an older PC. Just be patient.

When the software installation completes the DVD will squirt and the organisation will reboot. Be certain to grab the DVD out of the tray before the arrangement reboots to prevent the organisation from booting off the DVD again and wiping out the installation y'all just completed.

Afterward rebooting the installation routine pulls downward application updates. It says it'll take a couple minutes but it actually takes considerably longer fifty-fifty with a fast Internet connectedness so if yous have a slow connexion it'll take awhile.

Once the updates are complete you'll exist at the Linux

login: prompt. At that prompt type in root and hit Enter and and then type in the countersign that y'all entered during the software installation.

Screen shot courtesy of the AsteriskNOW Installation Page
Once you log in y'all'll be at what's called a "shell prompt" which is the Linux operating sytem waiting for you to enter a command. We'll be inbound commands afterwards just for now observe that just above trounce prompt is the "Interface eth0 IP" address. This is the IP address the system got via DHCP during the install procedure. (eth0 is Linux-ese for the PC's ethernet network adapter. Linux starts numbering things at zero.) This is the IP addres you'll apply to manage the system using the FreePBX GUI direction interface and enter into an H.323 configuration file.

Configure a Briefing Room

Go over to a Windows system that'due south on the same network as the Asterisk server, open a Web browser, and in the browser address line enter the Asterisk system's eth0 IP address nosotros just mentioned above. This will pull up the FreePBX direction GUI. This is where you have to enter that second countersign nosotros talked nigh earlier. The very first fourth dimension you go to this accost you'll be prompted to enter a user ID, countersign and admin east-postal service accost for the FreePBX application. As mentioned earlier, feel gratuitous to use root as the user name and the same password you lot used before if yous want.

One time an ID, countersign, and electronic mail address are entered a menu of applications is presented.

Screen shot courtesy of the AsteriskNOW Installation Folio
Click on the FreePBX Assistants icon and you lot'll be prompted to enter the user proper name and password that you just created. A System Overview folio will and so appear. (You tin can bring up this same page at any fourth dimension by pointing to Reports on the height menu bar and clicking on System Condition.)

The FreePBX menu is along the acme of the page. Point to Applications and click on the Conferences link to bring upwards the conference room configuration page.

Enter the phone number for the conference room and a descriptive name. The PIN numbers are optional. If you specify an Admin PIN and have the "Leader Await" option gear up to 'Yep' and then the callers volition not be able to communication with each other until the admin logs in. They'll be able to connect to the conference room but they'll simply hear music (if you set the "Music On Hold" pick to 'Yes') until someone with the Admin PIN logs in.

Set whatever of the other options you lot want. For instance, if the "User Join/Leave" option is set to 'Yes' the caller is asked to say their name when they log in and it is played dorsum into the room when they join and go out the call.

When using PIN numbers you lot want to let everyone know what the User PIN is just go along the Admin Pin a secret and only give it to those who will be conducting the conference calls.

Click on the Submit Changes button near the bottom of the folio and a red Apply Config push will appear adjacent to the menu at the pinnacle of the page. Click on this button to utilise the changes (which will reload the Asterisk application on the Linux arrangement) and then click on the Logout push button in the upper-right corner of the screen.

Nosotros're all washed with the Web-based GUI management tool. The next footstep is done back on the Linux system running Asterisk.

Create An H.323 Configuration File

Luckily the ooh323 channel driver for H.323 connectivity gets loaded automatically in AsteriskNow so all we have to practise is create a configuration file for the driver to use. Chances are likely the screen went into power salvage mode on the Linux system so just hitting the backspace key to bring it back.

At the shell prompt type in the following command and press Enter to get into the directory containing the Asterisk configuration files:

cd /etc/asterisk
If you desire to see how many configuration files in that location are type in ls and printing Enter. There are no H.323 configuration files and so we'll create one. At the trounce prompt blazon in the following command and press Enter to open a new, blank text file in the nano text editor.
nano ooh323.conf
and type in the following lines. Recall that when we outset logged in the system displayed the IP accost of the eth0 interface that yous used in the browser address bar to access the FreePBX GUI. Employ that same IP address for the bindaddr value. [general]
port=1720
bindaddr=10.24.66.68
disallow=all
allow=gsm,ulaw,alaw
dtmfmode=inband

[pbx]
type=friend
context=ext-meetme
host=172.20.1.18
port=1720

The 172.20.1.18 host entry is the IP accost of the PBX. The ext-meetme context is the context created by FreePBX for conference rooms.

Press Ctrl-Ten to exit and press the y (yes) central to ostend saving the file and printing Enter to confirm the file proper noun and you'll render the crush prompt.

We're all done configuring things on the Asterisk side so reboot the organization by typing in the following command at the trounce prompt:

shutdown -r now
If you ever desire to power down your PC intead of reboot it just replace the -r with -P (that's an upper-instance P). You never want to just turn off a Linux system.

Configuring the PBX

Naturally every PBX will be different on how you accomplish these, but the 2 tasks you need to complete are creating an H.323 torso and creating a route for the DN (phone number) you use for your conference room number (2663, which spells conf, in our example in a higher place). I'm using an Avaya IP Office system for examples on this page. While Avaya IP Office does come with conference bridge functionality it is very limited and is basically merely similar a Conference key on a telephone but allows more than participants.

Note that you lot should merely have to create a single H.323 trunk. If you plan on creating multiple conference rooms they each volition have their own DN (telephone number) and each DN will have it's ain route but each route will all utilise the same H.323 trunk (in the example of IP Office the same "Approachable Group ID").

On the left side of the Avaya IP Office system manager right-click on Line, point to New, and then click on H323 Line. On the VoIP Line tab the Line Number will be automatically assigned as will the Approachable Group ID. Yous can modify the Outgoing Group ID to a 3x, 4x or another value simply to make it more distinctive. The Telephone Number is more of a descriptive field for the sake of identifcation and inbound a number other than the one y'all used in the conference room properties volition not cause a problem. If you create more than ane conference room, i.due east. have more one telephone number using the H.323 link, you could list them all.

The two entries for aqueduct numbers will depend on the bandwidth of your IP connection between the PBX and the Asterisk organization. If information technology'due south all on the same LAN and utilization isn't high you could probably gear up information technology at the max (250) value. Continue in mind these are not per-conference-room limits merely a limit on all callers for all conference rooms that you volition create.

On the VoIP Setting tab for the Gateway IP Address you'll enter the same PBX address that you entered for the "host" value in the H.323 configuration file on the Asterisk organisation.

Y'all'll also want to change the Supplementary Services driblet-down to None. This will enable all of the check-boxes on the correct side of the window, some of which were previously greyed out. Now un-bank check all of these check-boxes. Recall that in the H.323 configuration file on the Asterisk system we ready the dtmfmode to inband which is why we accept to uncheck the Out of Band DTMF check-box.

We don't practise anything with short codes here and so y'all can now save this line configuration. On an Avaya IP Office organisation a simple line setup like this requires a system reboot but don't do that merely even so. Side by side nosotros'll add a brusk code to route the calls for 2663 telephone number to the H.323 line nosotros just configured.

On the left side of the IP Office system manager right-click on Short Codes and click on New. Here the Code is the phone number you lot used when y'all configured the briefing room. In the Phone Number field just put a menses. This means that the system will utilise the value (telephone number) that the user dialed, which will be the conference room number. Lastly, select the Line Group number that yous entered in the properties of the H.323 Line that you only created. Now save the brusque code and relieve the system configuration and you'll be informed that the system has to be rebooted.

If y'all want to create additional conference rooms all you demand to do is create the conference room on the Asterisk system using the FreePBX Web GUI and then add another short code on the PBX making sure the phone number for the new conference room is the same as the Code number in the new curt code. (Y'all'll use the same Line Grouping number in all additional short codes.) Past the manner, the phone numbers you utilise for each conference room are referred to every bit "extensions" in the Asterisk configuration files.

Testing and Troubleshooting

Once the PBX comes back up y'all should be able to place a telephone call to your briefing bridge telephone number and hear a prompt to enter a Pivot number (if you lot entered i in the configuration). If you don't, you'll want to come across if your calls are making it to the Asterisk system. Log into the Linux organisation with the root account and at the beat out prompt type in:

asterisk -rvvvvvv
Asterisk is already running as a process. The -r indicates that you want to connect to this process. The multiple v characters (enter 5 to 8 of them) indicates that you lot want a high level of verboseness, i.e. that yous want Asterisk to requite you as much feedback every bit possible. When you hit Enter yous'll exist in Asterisk's vanquish with a *CLI> prompt. This is Asterisk waiting for yous to enter a command but it will besides display feedback messages when an action is taken. (Blazon in quit when you want to exit the Asterisk CLI.)

With the Asterisk CLI up, telephone call the conference room number and run across if lines start scrolling on the CLI display. It'southward of import if lines do announced because it indicates that the telephone call is making it throughto your Asterisk arrangement. That would signal your PBX and Linux H.323 configurations are correct. This eliminates half the potential issues right there.

These lines will tell you lot what Asterisk is doing nigh the call and may requite you lot an error indicating what the problem is, particularly if it'south a syntax consequence considering y'all mis-typed i of the lines we entered earlier. Your best friend in this situation is Google. Enter any error exactly and within quotes then Google searches for information technology equally a continuous string. Some errors may not be anything service affecting but others may tell you lot what you need to correct.

If no text scrolls in the CLI interface double-check your ooh323.conf file for typos and if that's correct and then at that place'south likely an result with your PBX configuration.

Cut Off Announcements
A trouble I ran into was that the start of the voice prompts would get cut off. For instance, instead of hearing "Please enter the conference Pivot number" I would hear "rence PIN number." I discovered that this tin be remedied by adding a couple lines to the conference room configuration file on the Asterisk system.

All the same, editing the configuration files on a organisation running FreePBX is non a proficient idea because every time you make a change using the GUI and apply the changes these files get re-created so any changes yous brand are lost. That's why you'll see Asterisk conf files with "_custom" in the name. These files don't become recreated and it's where y'all'd normally want to put your custom changes. Unfortunately changing these custom files won't piece of work for us in this state of affairs and so if you lot need to add together these statements to fix an announcement issue, only practise so after y'all've created all of your conference rooms because you'll demand to add the corrective statements for each room (each extension).

On the Asterisk system enter the post-obit command to get into the directory with the Asterisk configuration files:

cd /etc/asterisk
So open the configuration file in the nano text editor with the command:
nano extensions_additional.conf
Page-downwardly the file and look for the Meetme context which starts with the following line:
[ext-meetme]
Below that you should encounter statement with your conference room(south) telephone number(southward) in them. Put the cursor at the beginning of the line that looks similar this:
exten => 2663,n,Read(Pin,enter_your_conference_pin_number,,,,)
and hit Enter twice to create two blank lines above it. Then enter the post-obit two commands on those blank lines using the approprite conference room phone number: exten => 2663,n,Respond
exten => 2663,due north,Expect(1)

Repeat this for any of your other conference room numbers in the [ext-meetme] context. When you're done press Ctrl-X to leave and printing the y (yes) key to confirm saving the file and printing Enter to confirm the file proper name and you'll render the vanquish prompt.

Now reboot the system by typing in the post-obit control at the trounce prompt:

shutdown -r now
Now when you call the conference room number you lot should hear the whole prompt. It has something to do with the VoIP line taking too long to get prepare and the statements we added forces information technology up faster.

Simply call back that if you ever brand any changes to your FreePBX configuration that file may go overwritten and you'll have to add together those statements again.

If you're a telecom worker check out my other page on edifice a

speak-back box using Asterisk. Keith'south Home Page

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