As Easy As 1-2-3: The Newbie’s Guide to TrixBox 1.2.3

NOTE: The system referenced in this article is no longer supported by Nerd Vittles as this version of Asterisk® has been phased out. For the latest and greatest, please consider our new PBX in a Flash offering.

Today we'll show you how to install the latest and greatest TrixBox 1.2.3 in about an hour. It is by far the best Asterisk-based IP PBX on the planet... especially once you add all of the Nerd Vittles goodies. It's been a painful couple of months in the TrixBox community, but the wait is over. Whether you're a casual home user or a gigantic call center processing millions of calls a month, this IP PBX can do it all reliably. And the best news: everything is FREE except the hardware on which to run your new system. For our new, automatic installation procedure, try PBX-in-a-Flash™. It's free, too!

What makes the TrixBox implementation really shine is freePBX 2.1.3 which brings to the table an incredibly simple yet powerful, upgradable web-based GUI to totally manage your PBX. And TrixBox adds all of the Asterisk bells and whistles you could ever ask for in an integrated PBX: full-featured database management, simple hooks to high-level application development tools such as PHP and Perl, an Apache web server, the MySQL DBMS, integrated voicemail and fax-to-email support, contact management, calling card billing, hardware autoconfiguration for Digium hardware as well as phone autoconfiguration for Cisco, Aastra, GrandStream, and Snom phones. In addition, you get built in Microsoft networking support, an integrated text-to-speech system, and loads of free utility software applications for Asterisk compliments of Nerd Vittles. And, yes, TrixBox 1.2.3 still fits on a single CD! For those new to Nerd Vittles, be aware that we make slipstream changes to articles as users discover things we've missed. Yes, we're human! So check for Comments before you begin or subscribe to our Comments RSS Feed. And, last but not least, be sure to add yourself to the Nerd Vittles Fan Club Map. So let's get started.

HOT TIP: For a turnkey version of TrixBox 1.2.3 that runs on your Windows desktop and includes the entire setup we'll be discussing as well as an out-of-the-box setup for 10 extensions and two VoIP providers, click here.

The Game Plan. Because of WordPress article length limitations and our own limited attention span, we're just going to cover the basics in this Guide. We'll leave a lot of the bells and whistles for future articles. We'll get your TrixBox system running so that you can make your first call. And we also will get your TrixBox system properly configured to support all of the dozens of Nerd Vittles' free applications.

Hardware Setup. You have two choices for hardware to run this new system. The first is to dedicate a machine to TrixBox and download the TrixBox ISO image to burn a bootable CD. Once you create the TrixBox CD, you simply boot your dedicated PC with the new CD. It will erase and reformat your hard disk for use with Linux and the included Linux and Asterisk applications. If you just want to experiment with TrixBox and don't plan to put the system into production other than for a few simultaneous calls, then you may prefer our PBX-in-a-Flash™ image. With this approach, you install VMware on your existing Windows XP system. Then you run Linux and the TrixBox application in a window on your Windows PC. It does not require a dedicated machine, and it's still FREE. We've found the performance to be virtually identical to running TrixBox on a dedicated PC provided your Windows machine has at least 512MB to 1GB of RAM. See our previous article for step-by-step instructions on the VMware installation process.

For today, however, we're assuming you've opted for the dedicated machine install: pure Linux on a clean machine. So begin by downloading the TrixBox ISO image from here and burn a CD (click here if you need a refresher course). Using your dedicated PC, insert the CD you made, plug your machine into the Internet, and turn it on. Then watch while TrixBox loads CentOS/4.4 (with an older Linux kernel that doesn't break voice applications!) and all the Asterisk and Linux goodies imaginable: Apache, SendMail, Asterisk Mail, SugarCRM, MySQL, PHP, phpMyAdmin, SSH, Bluetooth, freePBX, the Flash Operator Panel, Call Detail Reporting, and on and on. We've covered how to use most of the Linux products in our Mac HOW-TO's (see sidebar), and they work exactly the same way with TrixBox 1.2.3 so have a look around Nerd Vittles. There's plenty to keep you busy from now until Christmas. Remember! This install will reformat (aka ERASE) your hard disk before it begins, but it will warn you first. When you're prompted to create your root user password, type in something you can remember ... and write it down!

Upgrading TrixBox from a Prior Version of Asterisk@Home. In a nutshell, YOU CAN'T. But there is a way to put most of Humpty back together again once you've installed the new system. Before you begin, understand that you are doing this AT YOUR OWN RISK. NO GUARANTEES. If that bothers you, don't do it! The real trick is to do a little printing and copying of your old data before you insert that TrixBox installation disk. Step 1 is to make a full backup of your old system to a different server before you begin. If you don't know how, read our step-by-step instructions on the subject here. Step 2 is to make another copy of some of the critical files in your system. Duplicates of all of these will also be part of your backup. We typically build directories on a separate server which match the ones we'll be copying over from the old Asterisk system. Here are the directories (including all the subdirectories therein) that we always duplicate. Before you just blindly copy our list, stop and think whether there are special things you do on your existing Asterisk system or special apps that you run. Then find those files and make copies of all of them, too. The important piece in making a successful copy of some of these files is to shut down Asterisk (amportal stop) and MySQL (/etc/init.d/mysqld stop) before you begin. NOTE to CRM users: There's a new version of CRM in TrixBox so it's unlikely that you can restore the databases. Check your current version of AAH (help-aah) and see if there is an option (bundle-crm) to pack up CRM to move it to another machine. If so, do it and follow the instructions. We don't use Sugar so we haven't tested this upgrade option. Here are the directories you'll want to back up:

/var/lib/asterisk/agi-bin
/var/www/html
/var/lib/asterisk/sounds/custom
/var/lib/mysql
/root
/etc/asterisk

Then there are a couple of individual files that you'll also want to preserve:

/etc/hosts
/etc/crontab

The third step is to take screenshots of every screen you've built using the Asterisk Management Portal (AMP) or a prior version of freePBX. Start in the Setup tab and go right down the list of features. For each option in which you have multiple entries (e.g. Extensions and Trunks), call up each entry and print out the full page. Be especially careful in printing the Trunks entries and make sure you write down every line in the PEER Details and USER Details because those which are out of view will not get printed using a screen print. You'll need to manually fill in the ones that aren't displayed. The same goes for Registration Strings which often scroll out of view on the screen. Finally, using CLI (asterisk -r), make a copy of all your Asterisk database entries: database show. And, using phpMyAdmin, make a backup of any MySQL databases that you'd prefer not to lose, especially Airports, asteridex, sugarcrm, and mya2billing if you use and have added or changed data in any of those applications. Now save all this information in a safe place until we finish the new install.

Loading CentOS/4 and TrixBox 1.2.3. Here's how the install went for us, and we'll walk you through getting everything set up so that it can be used as a production server. Once the install begins, you can expect to eat up about 25 minutes with the CentOS 4.4 install. Just be sure to create your new root user password before you walk away, or it will still be sitting there waiting when you return. Once Linux is installed, the TrixBox CD will eject itself. You need to be present to remove the CD before the system reboots and begins the Asterisk 1.2.12.1 compile and install. That takes about 25 more minutes to complete.

Securing Your Passwords. When the install is finished and reboots again, log in as root with the password you assigned. Type help-trixbox for a listing of the other passwords that need to be changed. Change them all NOW!

passwd admin
passwd-maint
passwd-amp
passwd-meetme


Securing and Activating A2Billing. This web-based application allows you to generate and issue calling cards to individuals so that they can place calls remotely through your Asterisk server. If you've always wanted to be just like Ma Bell, here's your Big Chance! There's very little that you can do with an AT&T calling card that can't be done as well or better using A2Billing. And, it won't take an M.B.A. to undercut AT&T's calling card rates and still make buckets of money. All you need now are a few customers. But first, a word of caution. Assuming your Asterisk server has web exposure on the Internet, you need to secure the admin and root passwords in this application whether you use it or not. To access the application, go to http://192.168.0.104/a2billing/ using the actual internal IP address of your Asterisk server. Log in as root with a password of myroot. Click on the ADMINISTRATOR tab in the left column and then click Show Administrator. Now click on the Edit button beside each of the two administrator accounts and change the passwords to something secure. If you really would like to learn more about it, documentation for the application is available here. And, if you decide to use the application, you'll need to uncomment six actual dialplan lines in /etc/asterisk/extensions_trixbox.conf and reload Asterisk. Be sure to use a separate DID for this application and point it to custom-callingcard,s,1. We'll have a terrific new VoIP provider with dirt cheap DID rates to tell you about shortly so stay tuned!

;[custom-callingcard]
;exten => s,1,Answer
;exten => s,2,Wait,2
;exten => s,3,DeadAGI,a2billing.php
;exten => s,4,Wait,2
;exten => s,5,Hangup


Securing SugarCRM Contact Management. TrixBox includes the best open source contact management application on the planet, SugarCRM. You access the application with a web browser: http://192.168.0.104/crm/ substituting the private IP address of your Asterisk box, of course. We're going to stop repeating the substitution tip from here on. Whenever you see a reference to 192.168.something, just substitute the private IP address of your TrixBox server. Once the SugarCRM login screen appears, specify admin for your username and password for your password. Whether you use the application or not, change the admin password. It's easy. Just click the Administrator link under Welcome admin. Then click the Change Password button. Complete documentation for the application is available here. If contact management is your thing, knock yourself out, and we'll talk to you next spring when you finish getting everything set up to run your business. It's a great product, but be prepared to invest lots of time in the project if you expect to use it productively. If all you need is a phone dialer and a database that can display CallerID info for your friends and business associates, our AsteriDex product will be a better fit, and it takes about 2 seconds to enter each new person into the AsteriDex database.

If It Ain't Broke, Don't Fix It! Your new TrixBox includes a great utility to keep itself up to date. You simply log into your system as root and issue this command: trixbox-update.sh update. If the update script has also been updated, you'll need to run the command twice. That's the good news. The bad news is that these updates have broken TrixBox systems about as many times as they've fixed a problem. Our advice is to leave the update utility alone for the time being and only run it after someone you trust verifies that it won't bring your system down like a house of cards. The same applies to yum update. In earlier 1.2.x versions, this loaded a new Linux kernel which broke speech applications on many systems. What you'll have with TrixBox 1.2.3 is a very stable, secure system so don't break it by randomly adding new stuff. You've been warned.

Upgrading TrixBox to Support MailCall. The new TrixBox MailCall application needs POP3 and IMAP support for PHP in order to log into and read email messages to you over the phone using your email account. TrixBox 1.2.x versions are missing full PHP-IMAP support. Here's how to install what you'll need for versions 1.2 through 1.2.2. Log into your TrixBox system as root and issue the following commands in order.

cd /root
wget http://nerdvittles.com/trixbox122/libc-client-2002e-14.i386.rpm
wget http://nerdvittles.com/trixbox122/php-imap-4.3.11-2.8.i386.rpm
rpm -Uvh libc*
rpm -Uvh php*
apachectl restart

cd /var/www/html
wget http://nerdvittles.com/trixbox11/test.zip
unzip test.zip
rm -f test.zip


TrixBox 1.2.3 fixed part of the problem by including php-imap but left out the libc-client library so it still won't work without doing the following:

cd /root
wget http://nerdvittles.com/trixbox122/libc-client-2002e-14.i386.rpm
rpm -Uvh libc*
apachectl restart

cd /var/www/html
wget http://nerdvittles.com/trixbox11/test.zip
unzip test.zip
rm -f test.zip

To be sure everything went according to plan, use your web browser now to access http://192.168.0.104/test.php and be sure imap appears in the list of enabled libraries. Once you finish this tutorial, you should be all set to install the MailCall for Asterisk application.

Updating freePBX to the Latest Release. There's a more stable version of freePBX which isn't included in the TrixBox 1.2.3 build. Here's how to install it. Log into your server as root and issue the following commands:

cd /usr/src/
svn co https://svn.sourceforge.net/svnroot/amportal/freepbx/tags/2.1.3 freepbx-2.1.3
cd freepbx-2.1.3
./install_amp


Securing Asterisk. Because of a security vulnerability in the (Cisco) Skinny module of Asterisk, it needs to be disabled. Log into your server as root and edit modules.conf: nano -w /etc/asterisk/modules.conf. Then insert the following line in the [modules] context. Save the file and restart Asterisk: amportal restart.

noload => chan_skinny.so


Activating Bluetooth Support. Once the updates are completed, activate Bluetooth support if you plan to use it with our Follow-Me Phoning proximity detection application. Run setup, down arrow to System Services, press ENTER, down arrow to bluetooth and press the space bar, tab to OK, press ENTER, tab twice to Quit and press ENTER.

Activating Apache HTTPS Support. If you want secure Internet web access to your server, log into your system as root and issue these commands. Once https support is installed, you can access freePBX securely: https://AsteriskServerIPaddress.

yum -y install mod_ssl
shutdown -r now


Asterisk Info Application Is Back. One of the nice applications that previously was bundled in Asterisk@Home was Asterisk Info. It gave a detailed summary of many critical components in Asterisk including a listing of active SIP and IAX peers and registry entries. This is especially helpful when you're setting up new providers and want to see whether you're getting connected successfully. The application vanished in TrixBox 1.0, but it's back in TrixBox 1.1. You can run the application using a web browser pointed to the correct IP address of your server: http://192.168.0.104/. Then choose Asterisk Info from the TrixBox Configuration and Administration page. Or access it directly: http://192.168.0.104/maint/asterisk_info.php.

Simplifying SSH Access. If you're going to be connecting to other servers from your new TrixBox system using SSH or SCP, then build your new RSA key pair now. This lets you use SSH and SCP (secure copy) without having to enter a password each time. You can also automate backups and proximity detection scripts as we've explained previously here. Log in to your new TrixBox server as root. From the command prompt, issue the following command: ssh-keygen -t rsa. Press the enter key three times. You should see something similar to the following. The file name and location in bold below is the information we need:

Generating public/private rsa key pair.
Enter file in which to save the key (/root/.ssh/id_rsa):
Enter passphrase (empty for no passphrase):
Enter same passphrase again:
Your identification has been saved in /root/.ssh/id_rsa.
Your public key has been saved in /root/.ssh/id_rsa.pub.
The key fingerprint is:
1d:3c:14:23:d8:7b:57:d2:cd:18:70:80:0f:9b:b5:92 root@asterisk1.local

Now copy the file in bold above to your other Asterisk servers, Linux machines, and Macs. There's probably a way on PCs as well, but we've all but given up on that platform where security matters so you're on your own there. From your TrixBox server using SCP, the command should look like the following (except use the private IP address of each of your other Asterisk or Linux servers instead of 192.168.0.104). Provide the root password to your other servers (one at a time) when prompted to do so.

scp /root/.ssh/id_rsa.pub root@192.168.0.104:/root/.ssh/authorized_keys

On a Mac running Mac OS X, the command would look like this (using your username and your Mac's IP address, of course):

For user access only: scp /root/.ssh/id_rsa.pub wardmundy@192.168.0.104:/Users/wardmundy/.ssh/authorized_keys
For full root access: scp /root/.ssh/id_rsa.pub root@192.168.0.104:/var/root/.ssh/authorized_keys


Once the file has been copied to each server, try to log in to your other server from your new TrixBox server with the following command using the correct destination IP address, of course:

ssh root@192.168.0.104

You should be admitted without entering a password. If not, repeat the drill or read the complete article and find where you made a mistake. Now log out of the other server by typing exit.

Installing WebMin. We don't build Linux systems without installing WebMin, the Swiss Army knife of the Linux World. You can use it to start and stop services, check logs, adjust startup scripts, manage cron jobs, babysit your SendMail server, and many, many other tasks that are downright painful without it. If you ever need help from others, WebMin is a great tool for letting others help you.

There are lots of ways to install WebMin. WebMin now is part of the TrixBox yum repository so, after logging in as root, just issue the following command: yum -y install webmin.

WebMin runs its own web server on port 10000. To start WebMin, issue this command: /etc/webmin/start. You access it with a web browser pointed to the IP address of your Asterisk box (i.e. replace 192.168.0.108) at the correct port address, e.g. http://192.168.0.108:10000. Note, https support won't work on port 10000 without a bit of additional tweaking! The login name is root. Then type in your root password and press enter. The main WebMin screen will display. We really don't want the WebMin server starting up each time the OS reboots so do the following. Once you're logged in to WebMin, choose System->Bootup and Shutdown and then click on webmin. Click the No button beside Start at boot time, and then click the Save button. To stop WebMin when you're finished using it, issue this command: /etc/webmin/stop. You can restart it any time you need it, and then use a web browser to access it. But there's no need to waste processing resources. For complete WebMin documentation, click here.

[Note: This appears to be temporarily broken in the latest version of WebMin so try it again in a few days if it hangs during the Net::SSLeay install.] If you're going to be accessing WebMin from outside your firewall, you really don't want to be logging in as root over an unencrypted connection so let's enable https support for WebMin. While still logged into WebMin, click WebMin->WebMin Config->SSL Encryption. Now click Install Net::SSLeay Perl Module. Once the module is downloaded, click the Continue With Install button. The make and make install process will take a minute or two. Once you get the completed sucessfully message, click Return to WebMin. Choose WebMin->WebMin Config->SSL Encryption again. At the bottom of the form, click the Create Now button to create your SSL key. Click Return to WebMin again. Then choose WebMin->WebMin Config->SSL Encryption once more. Change the Enable SSL if available option to Yes, leave the other defaults, and save your changes. Henceforth, you can log into your server using HTTPS: https://TrixBoxIPaddress:10000/.

IP Configuration for Asterisk. We need a consistent IP address or domain name both on your internal network and externally if you expect to receive incoming calls reliably. There are three pieces to the IP configuration: (1) setting the internal IP address of your Asterisk server, (2) configuring a fully-qualified (external) domain name for your new server which will always point to your router/firewall, and (3) configuring your router to transfer incoming Asterisk packets to your Asterisk server. Here's how.

First, log into your server as root using your new password. Now type ifconfig eth0 (that's "e-t-h-zero") then enter, and write down both your inet addr and your HWaddr on the Ethernet 0 interface, eth0. Inet addr is the internal IP address of your Asterisk box assigned by your DHCP server (i.e. your router/firewall). HWAddr is the MAC address of your Asterisk server's eth0 network card. To assure a consistent internal IP address, you can either configure your router/DHCP server to make certain that it always hands out this same address to your Asterisk machine, or you can manually configure an IP address for this machine which is not in the range of addresses used by your DHCP server. Almost all routers now make it easy to preassign DHCP addresses so we prefer option 1. It's generally under the tab for LAN IP Setup or DHCP Configuration and is generally called something like Reserved IP table. Just add an entry and call it Asterisk PBX and specify the IP address and MAC address that you wrote down above. Now each time you reboot your Asterisk server, your router will assign it this same IP addreess.

To assure a consistent external address is a little trickier. Unless you have a static (fixed) IP address, you'll want to use a Dynamic DNS service such as dyndns.org and configure your router to always advertise its external IP address to dyndns.org. DynDNS.org will take care of revising the IP address associated with your domain name when your ISP changes your dynamic IP address. Then you can configure your VoIP provider account using your fully-qualified dyndns.org domain name, e.g. windswept.dyndns.org provides access to our beach house network even though Time Warner cable hands out dynamic IP addresses which change from time to time.

Now you'll need to log into your router and redirect certain incoming UDP packets to the internal IP address of your Asterisk machine. If you want external access to the Apache web server on your Asterisk machine, then map TCP port 80 to the internal IP address of your Asterisk system. For WebMin external access, map TCP port 10000 to your Asterisk system. If you want remote access to your Asterisk system via SSH, then map TCP port 22 to the internal IP address of your Asterisk system. If you want external IP phones or other Asterisk servers to be able to communicate with your Asterisk system, then map the following UDP port ranges to the internal IP address of your Asterisk system:

SIP 5004-5082
RTP 10001-20000
IAX 4569

For more details, read our full article on the subject.

Finally, you'll need to tell Asterisk about some of this. Edit the sip.conf file (nano -w /etc/asterisk/sip.conf) and add the following entries in the [general] section of the file using your fully-qualified domain name for your server and the private IP address range used behind your router/firewall (typically 192.168.0.0 or 192.168.1.0 with most home routers):

externhost = yourdomainname.dyndns.org
localnet=192.168.0.0/255.255.255.0
nat=yes


Designing Your PBX System. For those new to the Asterisk world, we'll be using a web-based GUI to configure Asterisk to meet your needs. Step 1 is to get away from your computer and sit down with a piece of paper. Now lay out how you'd like your new system to operate. How many phones will you have? Will they be software-based phones or good old phones you can put on a desktop? Will they be POTS phones (plain old touchtone phones), cordless POTS phones, SIP phones, IAX phones, or cordless SIP phones? How will you make and receive calls? Are you going to use an existing Ma Bell phone line or VoIP trunk lines from one or more VoIP providers? What should happen when incoming calls arrive? Do you want the caller to get an AutoAttendant message ("Hi. You've reached the Mundy's. Press 1 for Mary, 2 for Ward, or 3 to leave a message.") or do you just want all of your phones to start ringing? What should happen when no one answers or the line is busy? Do you want the calls transferred to a cell phone, another POTS phone, or just sent to voicemail? Which voicemail account? Should all busy phones send callers to the same voicemail account, or do you want one for each phone? What should happen once voicemail arrives? Do you want the phone to ring once a minute? Do you want the message waiting indicator to illuminate? Do you want the voicemail message to be emailed to you? Do you also want it preserved so that you can retrieve it from a touchtone phone? Do you want to be paged with the number of the person that called you?

ATTN: "Type A" Males. With apologies to our female readers, here's a tip for all the guys. If you have a wife (and want to keep her) or if you have teenage daughters (and want to avoid being killed in your sleep), you'd better get most of this PBX design right if you plan to use Asterisk to replace your existing home phone system. Otherwise, the day after you install your new system, a typical discussion with your spouse will begin with something like this: "What was wrong with our old phones that just rang when someone called and I could actually hear what they were saying when I answered?" With that caveat in mind, let's jump right in to freePBX.

Today's Objective. Keeping in mind that there are a million ways to configure and customize a PBX, we're going to walk you through a very simple setup today. Our objective is to get Asterisk and freePBX configured so that you can make a call and receive a call. In our next article, we'll start adding all the bells and whistles. But, for today, we'll show you how to set up an incoming and an outgoing VoIP trunk so you can make and receive free calls (at least in the U.S.) using a free softphone. When no one answers, the call will be sent to voicemail. And, when a voicemail message is left, the message will be emailed to you. We'll leave integration of existing POTS phones and phone lines for another day.

Choosing VoIP Providers. As you will quickly learn, choosing VoIP providers is an art, not a science. And it can be a slippery slope. A provider that is great one day can turn into an absolute nightmare the next. Take BroadVoice, for example. They used to be one of our favorites. Then the CEO left, and the company's business practices, uh, changed to put it charitably. You can read all about it on this forum or at the Better Business Bureau's site. All it takes is a change in leadership or direction at the company headquarters to go from first to worst overnight. So the best advice we can offer about choosing providers is this. Stay Flexible! Don't put all your eggs in one basket. And don't be in a hurry to disconnect your Ma Bell line and transfer your number until you are pretty confident about your provider. Six months is an absolute minimum, and a year is probably better. VoIP providers come and go at about the same pace as fast food restaurants in a new community.

Having said all of that, we have some providers we really like and some that we don't. YMMV! The basic idea in switching to Voice Over IP technology was to save money... not just for the provider, but for you, too. So PRICE MATTERS. There are typically three types of VoIP service: all-you-can-eat at a fixed monthly price, pay-as-you-go at a per minute (or part of a minute) rate, and free. Some providers only offer outbound service, and others offer incoming and outgoing calls. To receive calls, you've got to have an account with a provider that will give you a phone number unless you want to only get calls from other users of that provider's service, e.g. Skype. You don't have to use the same provider for inbound and outbound calls, and you are better off with backup providers for BOTH inbound and outbound calls.

If you select an all-you-can-eat plan, you basically get the right to make (or receive) ONE phone call at a time to a certain geographic area. This may be a state, an area code, or a country depending upon where you live and which provider you choose. The best of these in the U.S. is TelaSIP at $14.95 a month for unlimited US48 calling for the first three months, then $19.95. The runner-up is Axvoice which has a broader variety of plans including an unlimited international calling plan at $22.99 a month. Be aware of the fine print with all-you-can-eat providers. Some such as Teliax don't really offer unlimited calling even though they call it that. What they offer is unlimited calling up to some monthly cap of minutes. For example, with Teliax, up to 1500 minutes a month are "free" and then you pay 2¢ per minute thereafter. They're not really free because you've paid a $24.99 monthly fee for the initial 1,500 minutes. Then there's our old favorite BroadVoice which now offers unlimited calling with a little asterisk. After you drill down to the third level in their web pages, you'll see this in the fine print: "* Significant restrictions apply to Unlimited Plans." If you violate their undefined "normal residential usage patterns", you agree in advance to let them retroactively charge you 5¢ per minute for every call you've made since you signed up... plus $300/hour in in-house legal fees for successful collection. I wonder if they pay their staff attorneys that much? Their terms of use give them unfettered discretion in defining what's appropriate and inappropriate use. And, arguably, even having multiple people in your household use your "unlimited plan" violates their terms of service. So, unless you've recently won the lottery or just enjoy litigation, here's our best advice on BroadVoice: JUST SAY NO!

You'll also want to consider an incoming call provider that gives you one or more DIDs (phone numbers) in regions of some country or countries that you care about. Remember, with a local number, other local folks in that region can call you at no cost. This still matters. TelaSIP will give you a DID in the U.S. with free incoming calls for $5.95 a month. With AxVoice, it's $8.99 but you get three DIDs although there's no way to tell which one's ringing. Our new favorite for incoming calls is les.net at $3.99 a month in the U.S., and they've just added a slew of new cities (over 300 in the U.S. and Canada) which aren't yet shown on their web site. They also have terrific support for Asterisk and will have some more surprises in the next couple weeks.

And then there are pay-as-you-go providers for outbound calling. Usually there are no simultaneous call limitations because you're paying by the minute per call. Some of these providers charge in whole minute increments while others round calls to as little as six second billing increments. With some you get to specify your CallerID number while others always show the calls as Anonymous. Some leave their rates the same for six months or more. Others change their rates almost daily. You don't want to have to visit a web site each time your phone rings to determine what it will cost to pick up the phone. So be alert in choosing a pay-as-you-go provider. The best of the bunch in our opinion is Voxee.com at about a penny a minute for U.S. calls and only slightly more for calls to many international destinations.

Finally there are the (almost) free providers. Here's a good rule of thumb. Enjoy it while it lasts. Don't expect free or even almost free to last forever. And, most importantly, READ THE FINE PRINT. It costs the provider something to offer the service and, if they're giving the service away, there IS a catch. You just have to be smart enough to figure out what it is. The best freebies at the moment are VoipDiscount.com for free outbound calls to numerous countries including the U.S. for four months, FreeDigits.com for free incoming DIDs, free incoming calls, and free incoming fax service, and Stanaphone.com for free incoming DIDs and free incoming calls. See our complete list of VoIP Provider reviews for additional information and setup instructions.

If you just want to experiment with your new system and don't want to cough up much money, here's a good way to get your feet wet. Sign up for a free incoming DID number and free incoming calls with Stanaphone's Stana-IN service and sign up with VoIPDiscount.com for almost free outbound calls up to 300 minutes a week for four months. You'll need a Windows machine to initially sign up for both of these services. See our tutorials for details. You won't have a phone number in your local area code, but folks will be able to call you. If you want a number in your local area code and you live in the U.S., sign up for TelaSIP's basic service at $5.95 a month which gets you a local phone number and free unlimited incoming calls ... one at a time. Outbound calls in the U.S. are 2¢ a minute which gives you a good backup to your free VoIPDiscount outbound calling service. There are no obnoxious terms of service or hidden fees with TelaSIP. Just use the service for residential calling.

Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator or the snom 360 Softphone which is a replica of perhaps the best IP phone on the planet. Here's a new IAX softphone for all platforms that's great, too, and it requires no installation: Idefisk. All are free! Just install and then configure with the IP address of your TrixBox server. For username and password, use the extension number and password which we'll set up shortly with freePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best value in the marketplace with excellent build quality and feature set is the $85 GrandStream GXP-2000. It has support for four lines, speaks CallerID numbers, has a lighted display, and can be configured for autoanswer with a great speakerphone. Short of paying over double for the snom 360, that's as good as desktop phones get. If you want to use Asterisk throughout your home, buy a good 5.8GHz wireless phone system with plenty of extensions such as the Uniden 8866 which we use (see ad below) and then purchase an SPA-3000 to connect up both your home phone line and all your cordless phones. Our tutorial will show you how.

Initial Setup of freePBX. You still access freePBX just as you accessed the Asterisk Management Portal (AMP), by pointing a web browser to the internal IP address of your new Asterisk system. Once you get to the main TrixBox screen, choose System Administration and then freePBX. When prompted for your username and password, the username is still maint. Enter the password you assigned to freePBX/AMP when you configured your system. In the old days, AMP came preconfigured with everything they thought you'd need to use it. With the new freePBX architecture, you first have to install and enable the modules you want to use. And now others can write modules to expand the capabilities of freePBX without futzing around in the basic source code. You get to these modules by choosing Tools->Module Admin from the main freePBX menu. Unlike some applications, there's really no reason not to activate all of the available modules since they won't slow down Asterisk. The only performance hit is when you click the Red Bar to reload freePBX. The more modules you've activated, the longer it will take to reload freePBX (which isn't very long) since freePBX queries each module to see if changes need to be applied. So, in the Module Administration screen, click on each Module shown in the list and then Enable the Selected Modules. Now click Connect to Online Module Repository, select all the Modules in the list, and choose Download and Install the available modules. Repeat the drill one more time to get all of the modules from the Online Repository. Click the Red Bar to save your updates. From time to time, you need to revisit this page, connect to the online repository, and upgrade the modules in your freePBX system as new bug fixes are released.

As you can see, there are two types of Modules: Local Modules and Online Modules. Local Modules are the pieces that make freePBX work on your local machine. Online Modules provides access to modules which are available for download over the Internet. And Online Modules tells you which ones are newer than the ones currently on your system. HINT to Rob: Before too long, we wouldn't be surprised to see an option to email you notices when new modules are released or older ones are updated. freePBX is nothing short of fantastic for the Asterisk community if we do say so.

Last but not least, for each Module, there now is online documentation. You can read about all the Module pieces by clicking here. Once you complete the above steps, you're ready to set up your new system.

Configuring freePBX Trunks. When you click the Setup tab in freePBX, the first thing you'll notice is there are a lot more options. Start by adding your Trunks. This works pretty much like it always has. Choose ZAP, IAX2, SIP, or ENUM for each trunk and proceed accordingly. Down the road, the grand plan is to have sample settings for each provider on line here. Very cool!

For our sample setup today, we'll configure SIP trunks for Stanaphone, TelaSIP, and VoipDiscount. For each provider, click on the Setup->Trunks tab in freePBX. Then click Add SIP Trunk. After you complete the entries for each provider, click Submit Changes and then the Red Bar.

StanaPhone Trunk Setup. Here are the entries for the Stanaphone SIP trunk. For Outbound CallerID, enter the phone number assigned to you by StanaPhone. For Maximum Channels, enter 1. Leave the Dial Rules and Dial Prefix blank for the time being.

For Outgoing Settings, enter a Trunk Name of stanaphone. For Peer Details, enter the following using your assigned username and password. Be very careful to match the upper and lower case settings in your assigned password.

host=sip.stanaphone.com
insecure=very
nat=yes
secret=yourpassword
type=peer
username=yourusername


For Incoming Settings, enter a USER Context of from-pstn. This tells Asterisk to process incoming calls through this context in your dialplan. For USER Details, enter the following using your assigned username and password:

canreinvite=no
dtmfmode=rfc2833
host=sip.stanaphone.com
insecure=very
nat=yes
secret=yourpassword
type=peer
username=yourusername


For the Registration String, enter the following using your assigned username, password, and 347 phone number:

yourusername:yourpassword@sip.stanaphone.com/3471234567


Click the Submit Changes button and then click on the Red Bar to save your trunk settings and reload Asterisk. To be sure you have properly registered with Stanaphone, run the Asterisk_Info application which we installed above using your correct IP address: http://192.168.0.108/maint/asterisk_info.php. Under SIP Peers, you should see an entry for sip.stanaphone.com showing a state of Registered. If not, check your username and password entries for typos.

TelaSIP Trunk Setup. Here are the entries for the TelaSIP SIP trunk. For your Outbound Caller ID, fill in the local phone number provided by Telasip. For Maximum Channels, enter 1. For Dial Rules, enter the following:

1|NXXNXXXXXX
NXXNXXXXXX

In the Outgoing Settings section, name your trunk telasip-gw and then enter the following PEER details using your TelaSIP-assigned username and password:

context=from-pstn (if that doesn't work use: from-trunk)
dtmfmode=rfc2833
host=gw3.telasip.com
insecure=very
secret=yourpassword
type=peer
username=yourusername
fromuser=yourusername
sendrpid=yes

Leave the Incoming Settings User Context and User Details blank. For your Registration string, enter the following: yourusername:yourpassword@gw3.telasip.com using your actual username and password assigned by TelaSIP. Click Submit Changes and then the red bar to restart Asterisk. Use Asterisk_Info as we did with Stanaphone to be sure you are registering successfully with TelaSIP.

VoipDiscount Trunk Setup. Here are the entries for the VoipDiscount SIP trunk. Create a SIP trunk for the service with a Trunk Name of voipdiscount. VoipDiscount doesn't support an outbound CallerID number so leave it blank. The Outgoing Dialing Rules in the U.S. should look like this:

001+NXXNXXXXXX
00+1NXXNXXXXXX


Add the following PEER Details in Outgoing Settings using your own username (in three places!) and password. Leave the Incoming Settings blank.

allow=ulaw&alaw
authuser=yourusername
disallow=all
fromdomain=sipdiscount.com
fromuser=yourusername
host=sip.sipdiscount.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
sendrpid=yes
type=peer
username=yourusername


For the Registration String, enter the following using your own username and password:

yourusername:yourpassword@sip.sipdiscount.com

Click the Submit Changes button and click the Red Bar to update Asterisk. Use Asterisk_Info as we did with Stanaphone to be sure you are registering successfully with VoipDiscount.

When you have your Trunks set up, you'll need a way to call out (Outbound Routes), to call in (Inbound Routes), and to process incoming calls: a Digital Receptionist, a Call Queue, a Custom Application, DISA, or a phone to ring (Extensions). For today, we'll get the phones to ring. Then we'll tackle the other options in Parts II and III.

Configuring Outbound Routes. Outbound routes are the rules that determine how calls that are dialed from an extension on your system get processed. The idea here is that you set up a list of priorities. Then, based upon the number dialed, the outbound rules figure out how to route the call. We're going to start with a simple Outbound Route called Everything which will process all calls that are not handled by another Outbound Route. Click Setup->Outbound Routes->Add Route and enter the following:

Route Name ... Everything
Route Password ... [leave it blank]
Pin Set ... [leave it blank]
Emergency Dialing ... [leave it blank]
Dial Patterns: (adjust these if you wish to permit international calls!)
1NXXNXXXXXX
NXXNXXXXXX
Trunk Sequence:
0 sip/voipdiscount
1 sip/telasip-gw

Once you've made all the entries, click the Submit Changes button and then the Red Bar to reload Asterisk. You will be able to place calls by dialing either an area code and phone number or 1 plus an area code and phone number. For international callers, our previous articles will walk you through configuring the dial strings to support various countries. Now you should see two Outbound Routes in your route list. We want to delete the other route so just click on it and then choose Delete Route and click the Red Bar to save your changes. Now there should be only the Everything route in your Outbound Routes list. We'll leave it like that for today, but down the road, we'll add options for emergency calls, toll-free calls, in-state calls, and international calls. After we make those additions, the Everything route will be used as our lowest priority catch-all for calls that don't qualify for processing by another route.

Setting Up Extensions. To add a new extension and voicemail account to your system, click Setup->Extensions->Add SIP Extension and enter the following:

Extension Number ... 500
Display Name ... Office
Extension Options
Direct DID ... [your 10-digit TelaSIP phone number if you have one; otherwise, leave blank]
DID Alert Info ... [leave blank]
Outbound CID ... [your 10-digit TelaSIP phone number if you have one; otherwise, leave blank]
Emergency CID ... [your 10-digit TelaSIP phone number if you have one; otherwise, leave blank]
Record Incoming ... On Demand
Record Outgoing ... On Demand
Device Options
secret ... 1234
dtmfmode ... rfc2833
Voicemail & Directory ... Enabled
voicemail password ... 1234
email address ... yourname@yourdomain.com [if you want voicemail messages emailed to you]
pager email address ... yourname@yourdomain.com [if you want to be paged when voicemail messages arrive]
email attachment ... yes [if you want the voicemail message included in the email message]
play CID ... yes [if you want the CallerID played when you retrieve a message]
play envelope ... yes [if you want the date/time of the message played before the message is read to you]
delete Vmail ... yes [if you want the voicemail message deleted after it's emailed to you]
vm options ... callback=from-internal [to enable automatic callbacks by pressing 3,2 after playing a voicemail message]
vm context ... default


Configuring Inbound Routes. Just as we had to tell Asterisk how to process outbound calls, you also have to define what to do with incoming calls from each of your inbound trunks. Be aware that different service providers have implemented SIP and IAX differently. One of the best providers for proper SIP implementation is TelaSIP because you can route incoming calls based upon the DID numbers associated with each trunk. So you could have one incoming trunk from TelaSIP with multiple DID numbers (for each of your children, for example). Each DID then could be routed to a specific extension, and each extension could have its own CallerID number for outbound calls ... even though you might only have one TelaSIP trunk line. So, to outside callers, it would appear that each individual had his or her own phone line even though everyone might be sharing one or two trunks.

For today, we'll get a default inbound route established, and we'll save the gee whiz stuff for later. To create a Default Inbound Route for your calls, choose Setup->Inbound Routes->Add Route. Then enter the following:

DID Number ... [leave blank]
CallerID Number ... [leave blank]
Zaptel Channel ... [leave blank]
Fax Extension ... disabled
Fax Email ... [leave blank]
Fax Detection Type ... none
Pause After Answer ... [leave blank]
Privacy Manager ... no
Alert Info ... [leave blank]
Destination: ... Core: Office 500

Click Submit and then OK when you're warned that this will create a default incoming route for your calls. Down the road as you add additional incoming routes, the new routes will take precedence unless there's no matching DID in which case this default route will be used.

If you want to create a separate incoming route for your Stanaphone calls just to see how it works, click Add Incoming Route and enter the following:

DID Number ... [your 10-digit Stanaphone number]
CallerID Number ... [leave blank]
Zaptel Channel ... [leave blank]
Fax Extension ... freePBX default
Fax Email ... [leave blank]
Fax Detection Type ... NVfax
Pause After Answer ... 2
Privacy Manager ... no
Alert Info ... [leave blank]
Destination: ... Core: voice mailbox 500

The trick to learn here is that if you want an incoming DID to go straight to voicemail, you need a slight pause to let Asterisk get properly set up for the call or the first couple seconds of your voicemail announcement will be cut off. By adding two seconds of fax detection, everything will work swimmingly.

Allowing Anonymous Inbound SIP Calls. One final step, and your incoming calls should start arriving without a "this number is not in service" message. Choose Setup->General Settings and scroll to the bottom of the page. Under Security Settings, change Allow Anonymous Inbound SIP Calls from No to Yes and click Submit Changes and then the Red Bar. Once this change is made, inbound calls from Stanaphone will work reliably.

Activating Email Delivery of Messages. When you're out and someone leaves you a voicemail message or a fax, TrixBox and freePBX will let you forward that voicemail message or fax to your email address as a .wav file or PDF document. Or you can have the system send an instant message to your cell phone or pager telling you who called, what their phone number was, and how long a voicemail message the person left for you. Or you can do both. In addition, you can tell the system whether to delete the voicemail from your Asterisk server after sending it to your email account. In short, you now can manage all of your incoming email and voicemail from a single place, your email client.

You don't have to be a Linux expert to make this work. But you do have to master one concept. The Linux SendMail server imposes restrictions on who can send messages to eliminate the risk of spam. So ... THE SENDER DOMAIN FOR OUTBOUND EMAIL MESSAGES MUST MATCH THE SENDER DOMAIN OF YOUR ASTERISK SERVER AND YOUR SENDMAIL SERVER. Some Internet Service Providers also block email messages from downstream SendMail servers to reduce SPAM. Try the steps below to determine if the latter is a problem.

First, make this adjustment to the /etc/hosts file on your server. Edit the hosts file: nano /etc/hosts. Move the cursor to the second line which reads 127.0.0.1 asterisk1.local , and then move the cursor over the first letter of the first domain name shown, usually asterisk1.local. Now type in trixbox.dyndns.org and add a space after your entry. Don't erase the existing entry! Save your settings: Ctrl-X, y, enter. Now restart network services on your Asterisk machine: service network restart.

While not required for delivery of email messages, you'll still want to modify the email message which delivers your voicemails so that it includes IP address of your Asterisk system. You'll actually click on this link in the email messages to retrieve your voicemails over the web so make sure the address is Internet accessible. Also, don't do this using the TrixBox editor, or you'll mess up the formatting of the email message. You can download a fresh copy here if you need it. Instead, use nano: nano -w /etc/asterisk/vm_email.inc. Press Ctrl-W, type AMPWEBADDRESS, and press the enter key. Delete the word AMPWEBADDRESS and then type either the fully-qualified domain name for your Asterisk server or the private IP address if you only want to read your emails from behind your firewall. When you start typing, the text display may jump all over the place because of word wrap. Don't freak out. You haven't messed anything up. Once you complete your entry, don't erase or change anything else. Save the file: Ctrl-X,Y, then enter.

This step IS required to get emails delivered. Edit vm_general.inc: nano -w /etc/asterisk/vm_general.inc. Change the serveremail entry of vm@asterisk to vm@trixbox.dyndns.org. Then save your configuration and restart Asterisk: amportal restart. To get faxes delivered, adjust the Email From address of the faxes in freePBX->Setup->General Settings: fax@trixbox.dyndns.org. Note that these "from" addresses don't actually have to be working email addresses. They just need to match the domain that was entered in the hosts file and in the SendMail config file (below).

While still logged into your server as root, switch to the directory where the SendMail configuration file is stored: cd /etc/mail. Make a backup of the config file: cp sendmail.cf sendmail.cf.bak. Then issue the following command: echo CGtrixbox.dyndns.org >> sendmail.cf. Now restart SendMail on your server: /etc/rc.d/init.d/sendmail restart.

If your voicemails and faxes still don't get delivered to your email accounts, your ISP may be blocking downstream mail servers (that's you). Take a look at this link which will show you how to designate your ISP as your SMTP smart host using SendMail.

Activating the Nerd Vittles Weather Forecasts in TrixBox. TrixBox 1.2.3 includes the Flite text-to-speech engine as well as the Nerd Vittles weather forecasting system. To use it, just dial 611 from a phone on your system and enter a 3-character airport code to retrieve the weather forecast. TrixBox comes with support for about 50 airports. You can easily expand it to 1,000 airports by following along in Part II of our Weather Tutorial. It'll take you about 15 minutes. For complete instructions, read the full article here.

Activating Nerd Vittles CallerID on Steroids in TrixBox. We've previously written an application that will let your Asterisk PBX look up CallerID names for incoming calls from Google, and AnyWho, and your own AsteriDex database. The article explaining how to configure and install calleridname.agi is available here. And you can download the application here. Once you have installed and configured the AGI script, you can skip over the autoattendant changes mentioned in the article. Instead, make this simple change to your freePBX extensions.conf file. Point your brower to your TrixBox system and choose System Administration->Config Edit->extensions.conf. Then click on macro-user-callerid in the left column. Delete the existing contents and replace it with the following. NOTE: This only works with freePBX 2.1.3 so be sure you have that version installed before making the following change! Better yet, make a backup copy of extensions.conf first.

[macro-user-callerid]
exten => s,1,GotoIf($["${CHANNEL:0:5}" = "Local"]?report)
exten => s,n,GotoIf($["${REALCALLERIDNUM:1:2}" != ""]?start)
exten => s,n,Set(REALCALLERIDNUM=${CALLERID(number)})
exten => s,n(start),NoOp(REALCALLERIDNUM is ${REALCALLERIDNUM})
exten => s,n,Set(AMPUSER=${DB(DEVICE/${REALCALLERIDNUM}/user)})
exten => s,n,Set(AMPUSERCIDNAME=${DB(AMPUSER/${AMPUSER}/cidname)})
exten => s,n,GotoIf($["x${AMPUSERCIDNAME:1:2}" = "x"]?chknamelen)
exten => s,n,Set(CALLERID(all)=${AMPUSERCIDNAME} < ${AMPUSER}>)
exten => s,n(chknamelen),GotoIf($[${LEN(${CALLERID(name)})} > 0]?report)
exten => s,n,AGI(calleridname.agi)
exten => s,n(report),NoOp(Using CallerID ${CALLERID(all)})

; overrides callerid out trunks
; arg1 is trunk
; macro-user-callerid should be called _before_ using this macro

Click the Update button to save your changes and then reload Asterisk. Activate the services you wish to use in calleridname.agi making sure you've adjusted permissions on the file by following our previous tutorial, and you're all set.

Useful Functions on Your TrixBox System. Here's the complete list of functions that will work out of the box from any extension on your TrixBox system:

The Nerd Vittles Collection. We'll wrap it up for this week by pointing you to some tutorials for the Nerd Vittles applications that are preloaded in our VMware TrixBox build. You may also want to add these to your stand-alone TrixBox system. Each one takes less than 15 minutes to install.

AsteriDex - The Poor Man's Rolodex (http://TrixBoxIPaddress/asteridex/)

NewsClips for Asterisk - Get the News By Telephone (Dial 511)

MailCall for Asterisk - Get Your Email By Telephone (Dial 555)

Weather Reports by Airport Code - Get the Latest Weather Forecasts for 1,000 U.S. Cities (Dial 611)

Telephone Reminders for Asterisk - Appointment Reminders By Telephone (Dial 123)

GabCast Studio for Asterisk - Create and Play PodCasts Using Your Phone (Dial *422 and 422)

TeleYapper Message Broadcasting System - Deliver Appointment Reminders and Important Info to Any Custom Calling List (Dial 674)

Where To Go From Here. If you're new to the Asterisk world, you have lots of fun (and learning) ahead of you. After you finish this tutorial, you'll want to get plugged into the TrixBox Forums. That's the place to ask questions after you do some reading. Posting support questions on Nerd Vittles just doesn't work because of the cumbersome blog format. Don't email me questions either! We only accept thank you notes. Finally, take a look at our catalog of articles, projects, and Asterisk resource links. You'll find just about everything you'll ever need there. Enjoy!

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