Newbie’s Guide to TrixBox 1.0 and FreePBX 2.1.

1, Part I
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.

Well, the Nerd Vittles staff move is complete, and today we're back in the saddle. So, hello from
Charleston, South Carolina! And now there's a brand-new Asterisk@Home: TrixBox 1.0 with a brand-
new Asterisk Management Portal: freePBX 2.1.1. So we've got a lot of new ground to cover. These new
Asterisk products are designed to support the casual home or home office user's PBX needs as well as
gigantic call centers processing millions of calls a month. Everything is free except the hardware on
which to run your new system. That can be a $139 refurbished PC or a multi-processor RAID box with
mainframe horsepower. For home use, we've had great luck with older refurb units for under $150 each.
And, no, we're not on commission. How much commission could there be on this stuff? [Note: Updated
TrixBox 1.2.3 tutorial available here.]

What freePBX brings to the table is an incredibly simple yet powerful, upgradeable 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, integrated voicemail and fax-to-email
support, contact management, calling card billing, hardware autoconfiguration for Digium and Cisco
phone hardware, 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 1.0 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.

The Game Plan. Because of WordPress article length limitations and our own limited attention span,
we're going to divide this Guide into several parts. Today, we'll get your new system running so that you
can make your first call. In Part II, we'll cover a number of the bells and whistles that make TrixBox and
freePBX such a great combination. Then, in Part III, we'll add some more tips and tricks to help you
impress your friends whenever the need arises. And, no we haven't forgotten the other installments in our
weather report series. Our updated tutorial for TrixBox 1.1 is now available.

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 one or two simultaneous
calls from home, then you may prefer to download the VMware version of TrixBox or VMwarez's
enhanced version. With this approach, you install VMware on your existing Windows XP or Windows
2000 system. Then you run Linux and the TrixBox application in a window on your Windows PC. It does
not require a dedicated machine. 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 now, 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/4.3 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

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to use most of the Linux products in our Mac HOW-TO's (see sidebar), and they work exactly the same
way with TrixBox so keep reading. And, yes, this install will reformat (aka ERASE) your hard disk
before it begins, but it now warns you first. When you're prompted to create your root user password, type
in something you can remember ... or 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. Now save all this information in a safe place
until we finish the new install.

Loading CentOS/4 and TrixBox 1.0. 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. There is a wrinkle in the installation
process because of a Linux kernel upgrade which triggers a bug in Asterisk which triggers a missing
component in TrixBox, but we'll get all that fixed up in short order. Once the install begins, you can
expect to eat up about 25 minutes with the CentOS 4.3 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, reboot the system, and begin the Asterisk compile and
installation. That takes about 25 more minutes to complete.

Securing Your Passwords. When it's finished and reboots, log in as root with the password you
assigned. Type help-trixbox for a listing of the other four passwords that need to be changed. Change
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them all NOW!
passwd admin
passwd-maint
passwd-amp
passwd-meetme

Getting the Latest CentOS Updates. Once your system is secure, load all of the application updates for
CentOS 4.3. There now are lots of updates plus a new kernel install so be patient. If you have zaptel
cards, read this thread. The command to issue to begin the update process is yum -y update.

Rebuilding Zaptel. Every time there is a kernel update with yum (which is the case here), ZAP device
support needs to be rebuilt using the new kernel. Unfortunately, a RedHat bug caused the rebuilding
process to fail. Here's the fix. Log into your new server as root and issue the following commands to
determine which new kernel was loaded on your system:
cd /usr/src/kernels
ls

You should see the original kernel [Link]-something and the new one: [Link]-something.
Depending upon the processor in your system, the something may be different than our machine. Write
down the name of the new kernel directory and substitute it below for [Link]-i686. Now issue
these commands:
cd /usr/src/kernels/[Link]-i686/include/linux
mv spinlock.h [Link]
wget [Link]
shutdown -r now

In a perfect world, once the reboot completes, you should have been ready to rebuild ZAP device support.
But Andrew inadvertently left out the source code. So here's what you need to do next. Log into your new
system as root again and issue the following commands:
cd /usr/src
wget [Link]
tar -zxvf [Link]
mv zaptel-1.2.5 zaptel
cd /usr/src/zaptel
make clean
make install
shutdown -r now

Now we can rebuild support for your ZAP devices or ztdummy if you have no ZAP devices. Log in as
root again and type the following command: rebuild_zaptel. Then reboot your system: shutdown -r now.
Now log in as root again and type amportal stop and then genzaptelconf. Now, here's one final
housekeeping chore. Log in as root again and issue these commands:
touch /etc/[Link]
/usr/sbin/fxotune -s
shutdown -r now

Upgrading to Asterisk [Link]. Because of a serious security vulnerability in Asterisk, we are modifying
this article on June 17 to show how to load the Asterisk upgrade for those that followed this initial
tutorial. Log into your server as root and issue the following commands in order:
rpm -del [Link]
rpm -del [Link]
[Link]
[Link] update
reboot
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rebuild_zaptel
modprobe wcfxo [if you have zaptel hardware]
genzaptelconf
reboot

Now you should be good to go on the software front. Whew!

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: [Link]
yum -y install mod_ssl
shutdown -r now

Restoring Asterisk Info Application. 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. To restore the
application, log into your server as root and issue these commands:
cd /var/www/html/maint
wget [Link]
unzip asterisk_info.zip
rm -f asterisk_info.zip

Now you can run the application using a web browser pointed to the correct IP address of your server:
[Link]

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:
[Link] root@[Link]

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
[Link]). Provide the root password to your other servers (one at a time) when prompted to do so.
scp /root/.ssh/id_rsa.pub root@[Link]:/root/.ssh/authorized_keys

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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@[Link]:/Users/wardmundy/.ssh/authorized_keys
For full root access: scp /root/.ssh/id_rsa.pub
root@[Link]:/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@[Link]

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 [Link]) at the correct port address, e.g. [Link] 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.

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:
[Link]

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
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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 [Link] and configure your router to always
advertise its external IP address to [Link]. [Link] 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 [Link] domain name, e.g.
[Link] 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 [Link] file (nano -w
/etc/asterisk/[Link]) 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 [Link] or [Link] with most home routers):
externhost = [Link]
localnet=[Link]/[Link]
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

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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, let me chat privately for a moment with
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 U.S. calling. 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 tough 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,
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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!

With pay-as-you-go providers, there typically 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. 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 [Link] at about a penny a minute for U.S. calls and
only slightly more for calls to many international destinations.

And then there are the free providers. Here's a good rule of thumb. Enjoy it while it lasts. Don't expect
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 [Link] for free outbound
calls to numerous countries including the U.S. at least today, [Link] for free incoming DIDs, free
incoming calls, and free incoming fax service, and [Link] 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 [Link] for free outbound calls. 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 [Link] 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. The username
is still maint. Just 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
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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 clicking the freePBX option from the TrixBox main menu. Then choose
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 since it queries each module to see if changes need to be applied. So, in the
Module Administration screen, click Connect to Online Module Repository to first download all of the
available modules. Then select all of the Disabled Modules and Enable them. Click Submit and then the
Red Bar to save your updates. From time to time, you need to revisit this page to upgrade the modules as
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. 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. This 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=[Link]
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=[Link]
insecure=very
nat=yes

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secret=yourpassword
type=peer
username=yourusername

For the Registration String, enter the following using your assigned username, password, and 347 phone
number:
yourusername:yourpassword@[Link]/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: [Link]
Under SIP Peers, you should see an entry for [Link] 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=[Link]
insecure=very
secret=yourpassword
type=peer
username=yourusername

Leave the Incoming Settings User Context and User Details blank. For your Registration string, enter the
following: yourusername:yourpassword@[Link] 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=[Link]
fromuser=yourusername
host=[Link]
insecure=very
nat=yes
qualify=yes
secret=yourpassword
sendrpid=yes

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type=peer
username=yourusername

For the Registration String, enter the following using your own username and password:
yourusername:yourpassword@[Link]

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

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dtmfmode ... rfc2833
Voicemail & Directory ... Enabled
voicemail password ... 1234
email address ... yourname@[Link] [if you want voicemail messages emailed to
you]
pager email address ... yourname@[Link] [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 the next
chapter. 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

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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 VoiceMail Messages. When you're out and someone leaves you a
voicemail message, TrixBox and freePBX will let you forward that voicemail message to your email
address as a .wav file which can be played within most email client software. 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. In order to send out emails from your server, you'll need to make a few changes.

First, make this adjustment to the /etc/hosts file on the server. Since anonymous emails are blocked by
most ISPs, you'll need a fully-qualified domain name for your server. If you don't have your own domain,
the easiest alternative is to use the fully-qualified domain name that your ISP assigns to the IP address for
your broadband connection. Don't forget to update it when your ISP changes your IP address! To find out
what your fully-qualified domain name is, go to a command prompt on your Asterisk server and type:
nslookup [Link] substituting your public IP address for the preceding numbers. Then write
down the name entry without the trailing period. Now edit the hosts file: nano /etc/hosts. Move the cursor
to the second line which reads [Link] [Link] , and then move the cursor over the first letter of
the first domain name shown, usually [Link]. Now type in the fully-qualified domain name you
previously wrote down 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.

Next, you need to modify the email message which delivers your voicemails so that it includes your fully-
qualified domain name. Don't do this in TrixBox, 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.

Now edit vm_general.inc: nano -w /etc/asterisk/vm_general.inc. Change the serveremail entry of

vm@trixbox to an email name at the same fully qualified domain you used in your /etc/hosts file above.
Then save your configuration and restart Asterisk: amportal restart. If you continue with this setup and
still don't receive emails, here's another configuration change that is sometimes necessary. You'll also
need to do it if you reloaded settings from an older version of Asterisk. On the Asterisk terminal, log in as
root. Switch to the directory where the SendMail configuration file is stored: cd /etc/mail. Make a backup
of the config file: cp [Link] [Link]. Then issue the following command: echo
[Link] >> [Link]. Substitute the actual domain name of your Asterisk server for
[Link], but be sure it's preceded by CG with no intervening [Link] restart SendMail on
your server and try again: /etc/rc.d/init.d/sendmail restart. Finally, if your ISP doesn't permit downstream
mail servers (that's you), then take a look at this link which will show you how to designate your ISP as
your SMTP smart host using SendMail.

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Activating the Nerd Vittles Weather Forecasts in TrixBox. TrixBox now 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. We now
support about 50 airports. In our next installment, that will be expanded to 1,000 so stay tuned. For
complete instructions, read our original article.

Creating Wakeup Calls in TrixBox. To set up a wakeup call from any extension, dial *62 and enter a
two-digit hour and two-digit minute for the wakeup call.

Determining the Extension Number of Any Phone on Your TrixBox System. To determine the
extension number of any phone on your system, dial *65 from that extension.

Retrieving VoiceMail from Any Phone With TrixBox. To retrieve voicemail for any extension, dial
*98 and enter the voicemail extension number. When prompted, enter the password for that account. To
retrieve voicemail for the extension from which you are calling, dial *97 and enter the password for the
account when prompted. You can also set your voicemail defaults and record your voicemail greetings
using these options.

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:

 611 The Latest Weather Forecast

 *62 Schedule a Wakeup Call
 *65 Decipher Extension Number of Any Phone
 *70 Activate Call Waiting
 *71 Deactivate Call Waiting
 *72 Enable Call Forwarding (include forwarding number to avoid prompt)
 *73 Disable Call Forwarding
 *90 Enable Call Forwarding on Busy (include forwarding number to avoid prompt)
 *91 Disable Call Forwarding on Busy
 *78 Enable Do Not Disturb
 *79 Disable Do Not Disturb
 *97 Access Voicemail for Calling Extension
 *98 Access Voicemail with Prompt for Mailbox Number

Well, that should get you started. We'll tackle the gee whiz features in TrixBox and freePBX in our next
article so visit us again soon. In the meantime ...

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