How To Set Up A Teleconference
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Gratuitous Conference Bridge
A conference span allows multiple users to all dial the same telephone number and all be connected together every bit if they were in a virtual briefing room. A single briefing span can have multiple "rooms" with each room having it'due south 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 briefing calls can salve companies a lot of money in fourth dimension and travel and then they charge a premium for the ability to host such calls. The most mutual alternative to expensive PBX manufacturer options are hosted teleconference services with either monthly subscription charges or per-minute/per-user charges which can also go expensive.
Asterisk, the open source software PBX, has long offered MeetMe conference span functionality. The problem is getting it to interface with your corporate telephone system. 1 common option is to interface an Asterisk system to a PBX using a SIP trunk. The option nosotros'll use on this page is to set up up an H.323 connection between Asterisk and an Avaya IP Office PBX. However, it should work for whatsoever PBX that supports H.323 connections. Nosotros chose to use an H.323 connection because it is a more robust protocol (should you wish to expand beyond simple sound teleconferencing to teleconferencing with video and white lath capabilities at some indicate) and, on our phone arrangement, it didn't crave whatsoever additional licenses (we would have needed to purchase SIP licenses to use a SIP trunk).
There is an all-in-i version of Asterisk called AsteriskNow that includes not only the Asterisk PBX software simply the CentOS Linux operating arrangement and a Web-based GUI management interface for Asterisk chosen FreePBX. You lot just boot off of an AsteriskNow DVD and everything you need for a fully-functional PBX organisation gets installed and by and large configured for you.
Here's all you'll demand to practice to get a conference bridge working with an H.323-capable PBX:
- Download the free AsteriskNOW DVD ISO image and burn it to a DVD.
- Install AsteriskNOW on PC or server.
- Use the FreePBX GUI management tool to configure a briefing room.
- On the Asterisk server manually create an H.323 configuration file.
- On your PBX create an H.323 trunk.
- On your PBX create a short code (road).
Download AsteriskNOW
You can download the latest AsteriskNOW DVD ISO image from here. Naturally if you lot're going to exist using an older PC yous'll want to download the 32-fleck version.
world wide web.asterisk.org/downloads/asterisknow
Install AsteriskNOW
NOTE that installing AsteriskNOW will wipe out whatsoever is currently on the difficult-drive.
The DVD y'all created using the image you lot downloaded is called a "Asterisk distribution" considering it includes everything, the Linux operating system (CentOS), the Asterisk server awarding, the Apache Web server application to serve up the FreePBX Web pages, etc. etc. etc. Installing the software off of this one DVD will give you a consummate Asterisk PBX server.
While even an older PC volition work simply fine to set a test or small briefing bridge, if you lot program on setting up a production briefing bridge that will run across moderate to heavy use exist sure to take into account the maximum number of simultaneous callers you'll take on the system. The number of rooms or the number of callers in a room doesn't matter equally much as the number of all callers put together. If yous're going to accept fifteen or more people simultaneously dialed into the system y'all'll want to install the software on some pretty respectable server hardware with a fast CPU and a lot of memory.
Screen shot courtesy of the AsteriskNOW Installation Page
Boot off the DVD and the AsteriskNOW installation menu will appear. With Full Install under the newest version already highlighted printing 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 press the tab key to move the pocket-size highlight to the asterisk in forepart of "Enable IPv6 support" and press the space bar to de-select it. And then printing the tab key over again to highlight OK and press Enter.
Screen shot courtesy of the AsteriskNOW Installation Page
Likewise on the "Fourth dimension Zone Selection" screen, press the tab primal to highlight the list of time zones and use the up and down arrow keys to observe yours. You typically won't find "Eastern" or "Fundamental" and so you have to select a city (such as Chicago for Usa-Central). Once you have your fourth dimension zone city selected tab to OK and printing Enter.
Screen shot courtesy of the AsteriskNOW Installation Page
Next yous are prompted for a password. You will be prompted for two passwords during the installation and setup. This is the first one and information technology is for the Linux root account. On a Linux arrangement root is the super-user business relationship, similar Administrator is on a Windows machine. You need this password to log into the arrangement itself so don't forget what you enter here.
At this signal your work is done for the software installation part of the setup. All of the Linux, Asterisk, and FreePBX awarding packages volition exist installed automatically. The installation may seem like information technology hangs at various points, especially if you're using an older PC. Just be patient.The second password you'll be asked for is for the FreePBX GUI direction application. You'll not only be prompted for a password but for a user proper noun also. At that place's no reason you tin't employ "root" for that user name and the same password just to keep things unproblematic.
When the software installation completes the DVD will eject and the system volition reboot. Be certain to take hold of the DVD out of the tray before the system reboots to foreclose the organization from booting off the DVD once again and wiping out the installation you just completed.
After rebooting the installation routine pulls downwards application updates. It says it'll take a couple minutes but it actually takes considerably longer even with a fast Internet connection then if you accept a slow connection it'll accept awhile.
Once the updates are complete y'all'll be at the Linux
login: prompt. At that prompt type in root and striking Enter and so type in the countersign that y'all entered during the software installation.Screen shot courtesy of the AsteriskNOW Installation Page
Once y'all log in you'll exist at what'due south called a "trounce prompt" which is the Linux operating sytem waiting for you to enter a command. Nosotros'll be inbound commands later but for now notice that simply above vanquish prompt is the "Interface eth0 IP" address. This is the IP address the organisation got via DHCP during the install process. (eth0 is Linux-ese for the PC's ethernet network adapter. Linux starts numbering things at zippo.) This is the IP addres you'll use to manage the arrangement using the FreePBX GUI management interface and enter into an H.323 configuration file.
Configure a Conference Room
Go over to a Windows organisation that's on the same network as the Asterisk server, open a Web browser, and in the browser accost line enter the Asterisk system'due south eth0 IP accost we just mentioned in a higher place. This volition pull up the FreePBX direction GUI. This is where you have to enter that second password we talked most before. The very first time you get to this address you lot'll be prompted to enter a user ID, password and admin e-mail address for the FreePBX application. As mentioned earlier, experience free to utilise root equally the user proper noun and the same password you used earlier if you lot want.
Once an ID, password, and email address are entered a carte du jour of applications is presented.
Screen shot courtesy of the AsteriskNOW Installation Page
Click on the FreePBX Assistants icon and you'll be prompted to enter the user proper noun and countersign that you just created. A System Overview page will and then appear. (You lot can bring upwards this same page at any time by pointing to Reports on the top menu bar and clicking on System Status.)
The FreePBX menu is along the acme of the folio. Bespeak to Applications and click on the Conferences link to bring up the conference room configuration page.
Enter the telephone number for the conference room and a descriptive name. The Pin numbers are optional. If you specify an Admin Pin and have the "Leader Wait" pick set to 'Yeah' and then the callers will not be able to communication with each other until the admin logs in. They'll exist able to connect to the conference room but they'll merely hear music (if you set the "Music On Concur" pick to 'Yeah') until someone with the Admin Pivot logs in.
Ready any of the other options you want. For instance, if the "User Join/Go out" option is set up to 'Yeah' the caller is asked to say their name when they log in and information technology is played back into the room when they bring together and leave the call.
Click on the Submit Changes push nearly the bottom of the page and a blood-red Apply Config button volition appear next to the menu at the acme of the page. Click on this push to apply the changes (which will reload the Asterisk application on the Linux system) and so click on the Logout button in the upper-right corner of the screen.When using Pivot numbers you want to let everyone know what the User Pivot is but continue the Admin Pivot a secret and only give it to those who will be conducting the briefing calls.
We're all done with the Web-based GUI management tool. The adjacent step is done back on the Linux organization running Asterisk.
Create An H.323 Configuration File
Luckily the ooh323 aqueduct commuter for H.323 connectivity gets loaded automatically in AsteriskNow and so all we have to do is create a configuration file for the driver to utilize. Chances are likely the screen went into power salvage fashion on the Linux system and then just hit the backspace key to bring it back.
At the beat prompt type in the following command and press Enter to get into the directory containing the Asterisk configuration files:
cd /etc/asterisk
If you want to see how many configuration files there are blazon in ls and printing Enter. There are no H.323 configuration files so nosotros'll create one. At the crush prompt type in the following command and printing Enter to open a new, bare text file in the nano text editor.
nano ooh323.conf
and type in the following lines. Recall that when we first logged in the system displayed the IP address of the eth0 interface that you used in the browser address bar to access the FreePBX GUI. Use that aforementioned IP address for the bindaddr value. [general]
port=1720
bindaddr=ten.24.66.68
disallow=all
allow=gsm,ulaw,alaw
dtmfmode=inband
[pbx]
type=friend
context=ext-meetme
host=172.twenty.ane.18
port=1720
The 172.xx.1.xviii host entry is the IP address of the PBX. The ext-meetme context is the context created past FreePBX for conference rooms.
Press Ctrl-X to leave and press the y (yes) central to confirm saving the file and press Enter to confirm the file name and you'll return the trounce prompt.
We're all done configuring things on the Asterisk side so reboot the system by typing in the following command at the crush prompt:
shutdown -r at present
If y'all always want to power down your PC intead of reboot it simply replace the -r with -P (that's an upper-case P). You never want to just turn off a Linux organisation.
Configuring the PBX
Naturally every PBX will be different on how you accomplish these, simply the two tasks you need to consummate are creating an H.323 torso and creating a route for the DN (telephone number) y'all employ for your conference room number (2663, which spells conf, in our example higher up). 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 express and is basically just like a Conference cardinal on a phone only allows more participants.
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 volition be automatically assigned equally will the Outgoing Group ID. Y'all can change the Outgoing Grouping ID to a 3x, 4x or another value only to go far more distinctive. The Telephone Number is more of a descriptive field for the sake of identifcation and entering a number other than the 1 yous used in the conference room properties will not cause a trouble. If you lot create more than than one conference room, i.e. have more than 1 telephone number using the H.323 link, you could list them all.Notation that you should only have to create a single H.323 torso. If you plan on creating multiple conference rooms they each will have their ain DN (telephone number) and each DN volition have it'due south own route simply each route will all use the aforementioned H.323 body (in the case ofIP Role the aforementioned "Outgoing Group ID").
The two entries for channel numbers will depend on the bandwidth of your IP connection between the PBX and the Asterisk system. If it's all on the same LAN and utilization isn't high you lot could probably prepare information technology at the max (250) value. Go along in mind these are not per-briefing-room limits but a limit on all callers for all conference rooms that yous volition create.
On the VoIP Setting tab for the Gateway IP Accost you'll enter the same PBX accost that you entered for the "host" value in the H.323 configuration file on the Asterisk arrangement.
You lot'll also want to change the Supplementary Services drib-downwardly to None. This volition enable all of the check-boxes on the right side of the window, some of which were previously greyed out. Now un-check all of these check-boxes. Recall that in the H.323 configuration file on the Asterisk system we prepare the dtmfmode to inband which is why nosotros have to uncheck the Out of Band DTMF cheque-box.
We don't practice anything with curt codes hither then you can now save this line configuration. On an Avaya IP Office system a elementary line setup like this requires a system reboot but don't do that merely even so. Adjacent we'll add a curt lawmaking to route the calls for 2663 telephone number to the H.323 line nosotros just configured.
On the left side of the IP Office arrangement managing director right-click on Brusk Codes and click on New. Here the Code is the phone number y'all used when you configured the briefing room. In the Telephone Number field merely put a period. This means that the system will utilize the value (phone number) that the user dialed, which volition exist the conference room number. Lastly, select the Line Group number that yous entered in the properties of the H.323 Line that you simply created. Now save the short code and salve the system configuration and you'll be informed that the system has to be rebooted.
If yous want to create additional conference rooms all you need to do is create the briefing room on the Asterisk arrangement using the FreePBX Web GUI so add some other short lawmaking on the PBX making certain the phone number for the new conference room is the same every bit the Lawmaking number in the new brusque code. (You'll use the same Line Group number in all additional short codes.) By the way, the phone numbers you utilize for each conference room are referred to as "extensions" in the Asterisk configuration files.
Testing and Troubleshooting
In one case the PBX comes dorsum up y'all should be able to place a call to your conference bridge telephone number and hear a prompt to enter a Pin number (if y'all entered ane in the configuration). If you don't, you'll want to see if your calls are making information technology to the Asterisk arrangement. Log into the Linux system 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 want a high level of verboseness, i.eastward. that you want Asterisk to requite you every bit much feedback as possible. When you hit Enter y'all'll exist in Asterisk's crush with a *CLI> prompt. This is Asterisk waiting for you to enter a control but information technology will also display feedback messages when an activity is taken. (Type in quit when yous want to exit the Asterisk CLI.)
With the Asterisk CLI up, telephone call the conference room number and see if lines start scrolling on the CLI display. It's important if lines do appear because it indicates that the call is making it throughto your Asterisk system. That would indicate your PBX and Linux H.323 configurations are correct. This eliminates one-half the potential issues correct at that place.
These lines will tell you what Asterisk is doing well-nigh the call and may requite you an error indicating what the trouble is, specially if it's a syntax issue because you lot mis-typed one of the lines we entered before. Your all-time friend in this situation is Google. Enter any fault exactly and within quotes so Google searches for information technology as a continuous string. Some errors may not be anything service affecting but others may tell you what you need to correct.
If no text scrolls in the CLI interface double-cheque your ooh323.conf file for typos and if that's correct so at that place'south probable an issue with your PBX configuration.
HOWEVER, editing the configuration files on a organisation running FreePBX is not a practiced idea because every fourth dimension you make a change using the GUI and apply the changes these files get re-created so any changes you brand are lost. That's why you'll run into Asterisk conf files with "_custom" in the name. These files don't become recreated and it's where you lot'd ordinarily desire to put your custom changes. Unfortunately changing these custom files won't work for united states in this state of affairs so if you need to add these statements to fix an proclamation outcome, only practice and so after you've created all of your conference rooms because you'll need to add the corrective statements for each room (each extension). On the Asterisk system enter the following command to get into the directory with the Asterisk configuration files: At present reboot the system by typing in the following command at the shell prompt: Just recall that if y'all ever make whatsoever changes to your FreePBX configuration that file may become overwritten and you'll have to add those statements over again.
A problem I ran into was that the offset of the phonation prompts would get cut off. For example, instead of hearing "Please enter the conference PIN number" I would hear "rence Pivot number." I discovered that this tin can be remedied by calculation a couple lines to the briefing room configuration file on the Asterisk system.
cd /etc/asterisk
And then open up 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]
Beneath that you should see statement with your conference room(south) phone number(due south) in them. Put the cursor at the beginning of the line that looks like this:
exten => 2663,n,Read(Pivot,enter_your_conference_pin_number,,,,)
and striking Enter twice to create 2 blank lines above it. So enter the following ii commands on those blank lines using the approprite conference room telephone number: exten => 2663,n,Answer
exten => 2663,due north,Look(1)
Echo this for whatsoever of your other conference room numbers in the [ext-meetme] context. When you're done press Ctrl-10 to leave and press the y (yes) key to confirm saving the file and press Enter to ostend the file proper noun and you lot'll return the shell prompt.
shutdown -r now
Now when you phone call the briefing room number you should hear the whole prompt. It has something to do with the VoIP line taking likewise long to get set up and the statements we added forces it up faster.
If you're a telecom worker check out my other page on building a
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How To Set Up A Teleconference,
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