Newbie’s Guide to TrixBox 1.1 and freePBX

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 1.1 with freePBX 2.1.1 in just over an hour. As with the earlier release of TrixBox, 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 almost any old Pentium PC or a multi-processor RAID box with mainframe horsepower.

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 1.1 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.

UPDATE: This Guide has been superceded. For the TrixBox 1.2.3 tutorial, 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. So today we'll get your TrixBox 1.1 system running so that you can make your first call. We also want to get TrixBox properly configured to support our next free application: TrixBox MailCall. It'll let you retrieve and play back your email messages using any touchtone telephone and your TrixBox system. And, yes, you'll need TrixBox 1.1 to make everything work. The latest TrixBox 1.1.1 update (covered below) will get any system properly configured for the MailCall for Asterisk application. Thanks, Andrew!

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 1.0 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. And note that there isn't yet a VMware version of TrixBox 1.1 so follow the Newbie's Guide to TrixBox 1.0 to get everything working if you go the VMware route. TrixBox MailCall will not work with TrixBox 1.0 so, if that's of interest to you, install TrixBox 1.1. Once you run the trixbox-update.sh script twice (covered below), you'll have the 1.1.1 version running under VMware.

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 to use most of the Linux products in our Mac HOW-TO's (see sidebar), and they work exactly the same way with TrixBox 1.1 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 ... 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. Now save all this information in a safe place until we finish the new install.

Loading CentOS/4 and TrixBox 1.1. 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 4.3 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 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 AT&T, 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 by you 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 extensions_trixbox.conf and reload Asterisk. Be sure to use a separate DID for this application and point it to custom-callingcard,s,1.

;[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. 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.

Getting the Latest TrixBox Updates. Once your system is secure, load all of the TrixBox updates using one simple command. Log into your TrixBox 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.

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 from your email account. The latest TrixBox update adds everything you'll need to either TrixBox 1.0 or TrixBox 1.1 installs. Currently, the two libraries to support this aren't included in TrixBox so here's how to install them. Log into your TrixBox system as root and issue the following commands in order:


cd /root
wget http://nerdvittles.com/trixbox11/libc-client-2002e-14.i386.rpm
wget http://nerdvittles.com/trixbox11/php-imap-4.3.9-3.9.i386.rpm
rpm -Uvh libc*
rpm -Uvh php*

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

Reconfiguring Apache to Support PHP. At least on our system, TrixBox 1.1 was misconfigured for PHP applications to function properly with Apache. Note: This may have been fixed in the 1.1.1 update so, after downloading the test.zip file above, test your system by executing this command from a web browser using the actual IP address of your TrixBox system instead of our IP address: http://192.168.0.129/test.php. If you get a pretty PHP display about your system, you can skip the next step. If you just see three lines of code or nothing at all, then do the following while still logged in as root:

cd /etc/httpd/conf
cp httpd.conf httpd.conf.bak
nano -w httpd.conf

Once the editor opens your Apache config file, press Ctrl-W and search for the following: LoadModule access_module. After pressing Enter, move to the left margin of that line, and press Enter to open up a blank line. Insert the following code above the existing LoadModule access_module line:

LoadModule php4_module modules/libphp4.so

Now press Ctrl-W again and search for the following: AddType application/x-tar. After pressing Enter, open up a blank line below the existing entry and insert the following:

AddType application/x-httpd-php-source phps
AddType application/x-httpd-php php

Finally press Ctrl-W a third time and search for the following: #AddHandler cgi-script. After pressing Enter, add the following code below the existing entry:

AddHandler php-script php

Save your changes by pressing Ctrl-X, then Y, then Enter. Restart Apache to activate the changes: /etc/init.d/httpd restart. Now run the test.php script from your web browser again, and you should be all set.

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://192.168.0.129/. Then choose Asterisk Info from the TrixBox Configuration and Administration page.

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.

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, 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, 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 Voxee.com 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 VoipDiscount.com for free outbound calls to numerous countries including the U.S. at least today, 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 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 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 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 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=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

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 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 123.456.789.001 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 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 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 sendmail.cf sendmail.cf.bak. Then issue the following command: echo CGasterisk.dyndns.org >> sendmail.cf. Substitute the actual domain name of your Asterisk server for asterisk.dyndns.org, but be sure it's preceded by CG with no intervening spaces.Then 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.


Activating the Nerd Vittles Weather Forecasts in TrixBox. TrixBox 1.1 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. 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.

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 1.1 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 down the road so visit us again soon. In the meantime ...


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    29 Responses to “Newbie’s Guide to TrixBox 1.1 and freePBX”

    1. Thanks once more for yet another great free app! I really wish you hit the lottery or something ’cause you deserve it! Anyway, uhh… How do I get the system to read my email though?

      [WM: Check back next week. With a little luck, we’ll have everything working and ready to distribute. And … great suggestion on the lottery.]

    2. Paul Rocha says:

      Thanks again, where would we be without your help. I find myself almost addicted to your blog. I think I actually get the shakes when nothing new is posted for a while ;o). Ward, after the install I get a failed status when trixbox is restarting (no zaptel modual found). Any ideas what could have gone wrong?

      [WM: Andrew just released a huge new update this morning so run trixbox-update twice and see if that doesn’t get things squared away. If all else fails, try the tricks in the TrixBox 1.0 tutorial to restore Zaptel functionality.]

    3. Micronet says:

      Thanks for your work. I visit your website every week to learn new lessons. In the trixbox forum, they talked about php 4.3.9 does not work well with SugerSRM, need to upgrade to PHP4.4.2, could you please give a detailed instruction how to do it? Thank you again.

      [WM: Got this note from Andrew this morning.]

      “Also I just added php-imap to a new 1.1.1 upgrade. just type “trixbox-update.sh update” and you will get all the latest stuff + php-imap. (This works on any version). I have also updated PHP to 4.3.11 This should fix all the SugarCRM problems and some other PHP issues. This auto-upgrade stuff was a pain to set up but now that it’s working it’s really cool!”]

    4. ward says:

      Just a heads up that Andrew released the TrixBox 1.1.1 update (through the trixbox-update) after this article was published yesterday. This is a must-have upgrade that solves the issues with PHP for both our upcoming MailCall application as well as SugarCRM. Just run “trixbox-update.sh update” twice to get everything loaded including a new Linux kernel. Then reboot.

    5. Maceo says:

      Ward,

      Excellent tutorial, as always. Quick question, I have been using my Sipura SPA-3000 for the entire time that I have played with Asterisk. I starting fiddling around with VoIPDiscount to test VoIP calls.

      When I talk to people on that trunk instead of the PSTN, people say that when I start talking, my first syllable on my words gets cut off..similar to what happpens sometimes when you talk on a speakerphone. Is this a provider specific issue? Are there providers where this doesn’t happen? Or, is this a codec thing and if I buy a license for the Digium codec will that fix it?

      Thanks again for sharing your knowledge.
      Maceo

      [WM: You’ll get a better answer by posting this on either the TrixBox or Voxilla forum.]

    6. Mike says:

      Ward – where are you?? We need more. Hope you have settled in after the big move!!! Thx for all of the help!!

      [WM: All settled. We’re at pawleys.org until tomorrow. MailCall for Asterisk (listen to your email from any phone!) also coming tomorrow with a little luck.]

    7. Ian says:

      I am running trixbox 1.1.0 behind a NAT firewall. I signed up for a voipexpress account from a single incoming DID. I followed your example for the setting up the trunk. It only worked after I changed the context from context=from-internal to context=from-trunk. With the context set to from-internal I would just get a not available tone and the asterisk responds SIP/2.0 404 Not Found. Not sure if this is a bug or a feature.

    8. Roberto Ocampo says:

      I have record in some mp3 files with all of the sounds from a list of text for the IVR for Asterisk. File loaded from Digium web site. But I don’t know what to do next. Can you tell me a way to convert them to a g729. I have read that this is the better codec. Is it posible? The list of texts were in English and i’ve translated them to Spanish-Mexico. I thik i will be a good contribution. Can you tell me a way?

    9. Iain says:

      Wondeful tutorials, I find them very interesting and easy to follow.

      A question: what is the most stable version of Asterisk@Home/Trixbox?
      I prefer stability to leading edge.

      Asterisk 1.5 used to be great
      What about the 2.x series of Asterisk 2.7, 2.8
      which one has most things that work :)

      Thank you again Iain

      [WM: All of the AAH and TrixBox versions were built on top of each other. At this point, it’s an easy call. TrixBox can be automatically updated. Asterisk@Home can’t. So, for once, the bleeding edge is probably the most stable.]

    10. Keen says:

      Hi, great tutorial! I’m still running an old A@H, will be upgrading once I find the time to. However there’s something that hangs at the back of my head. Currently I have 1 X100P clone FXO card, and I’m thinking of getting another FXO, so that I can dial home from my mobile, and initiate the trixbox to callback to either give me a prompt to use the 2nd FXO to call out to any landline/mobile, or some way of telling the box to dial a number and callback to connect both sides. Is this possible? Would you be able to write a guide/tutorial on this?

      Many thanks for your response in advance.

      Thank you.

      [WM: There are a number of articles already. Just search this site for DISA.]

    11. John says:

      Incredible! So much work for a free product….. Incredible.
      Oops.. the “install-pdf” doesn’t work now… I can’t find any info in the forums, but maybe I’m not looking hard enough.

    12. Tom says:

      Don’t forget that the Sugar CRM has an admin account with the password “password”. There are probably several others that we need to find, since you don’t want the bad guys getting any access to exploit your box.

    13. Maceo Woodward says:

      Does anybody know why the text coloring that existed in Asterisk@Home was ditched in Trixbox? Just curious…

      Maceo

    14. Daniel says:

      You may want to change your command to copy the ssh keys to the following:
      cat ~/.ssh/id_rsa.pub | ssh xxx.xxx.xxx.xxx "cat >> ~/.ssh/authorized_keys"
      If you use the code you have listed, it will overwrite the previously authorized keys, causing confussion, heartburn, and possibily headaches.

    15. Sean says:

      I am running trixbox 1.1.1 and followed the setup above. Most everything works except the Linksys PAP2T-NA. I have a uniden phone plugged into it and I can make calls out. When a call come in th euniden does not ring and does not show caller id. I hear another phone ringing so I pick up the uniden and the call is there and I can talk fine. So it looks like the pap2t is not sending anything over to the uniden. The amount of options in that thing is crazy so I really have no idea where to even start. Has anyone seen this before?
      Thanks
      Sean

    16. Scott says:

      Thanks for all the great articles. I have been using Telasip for 8 months now and love them. One quick note to clear any confusion…Telasip uses multiple gateways so the hostname may not be gw3.telasip.com. It might be gw4.telasip.com, gw5.telasip.com, gw6.telasip.com, etc. Regardless of the gateway, their support staff will tell you which gateway your DID is assigned to use.

    17. summers311 says:

      Hey Ward! Things are a bit weird here. I guess there was another kernal update and it makes things a little screwy. Its 3:00 in the morning on 8/4 here and I am setting up trixbox at my company and am faithfully following your directions as usual, what happens is when you log in the first time it gives you the option of 2 kernals I think one that ends EL and one ELsmp, it defaults to ELsmp and if you follow your instructions in ELsmp it will not load zaptel no matter what you do, but if you make sure to stay in EL and do not do yum updates just trixbox-update.sh I was fine, just thought I would give people a heads up on this and would love to know what the difference is between EL and ELsmp if you know. Can?ǨѢt wait until your next post hope you update TeleYapper soon im a big fan of that one?Ǩ.

      [WM: Multiprocessor machines (SMP) still are a problem. Andrew doesn’t have one to test. Updates for all our apps are coming … starting next week with AsteriDex.]

    18. Frank Daley says:

      Awesome intro to Trixbox.

      Now trying to get fax-to-email working. Anyone know of a guide for Trixbox 1.1?

      ta
      frank

    19. tphank says:

      Ward: In the past you instructed to do the “yum -y update” after the trixbox update. Should this still be the case?

      [WM: The TrixBox update script now takes care of this.]

    20. lockdog says:

      great intro.. trying to connect a voiceblue cellular/VoIP gateway as a trunk.. having registration problems.. getting SIP/404 errors.. can sometimes get in as an unknown peer.. any ideas

    21. Andrew Mc says:

      I have Sirrix ISDN cards which require the /usr/include/asterisk folder. Trixbox uses CVS code which doesn’t have these folders and the driver install fails. AAH 2.8 looks to use regular release code which does. I don’t know who to tell to look at having these folders put back in the build, but you seem to know so perhaps you could tell them for me.
      I have tried puting them in and making some links (trixbox forum) but the update script breaks it again.

    22. Tricky says:

      Is there anyway to tie it in with vonage? Or do i need to put a digium card in the pc?

      Thanks in advance,

      Tricky

      [WM: Just get an SPA-3000 and search this site for the tutorial.]

    23. Frank says:

      Ward,
      I did as you instructed us to do. I clicked to goto the Online Module Respository and this is what I got?

      Terminate Connection to Online Module Repository
      Warning: file_get_contents(http://amportal.sourceforge.net/modules-2.1.xml): failed to open stream: No route to host in /var/www/html/admin/page.modules.php on line 445

      What do I need to do?
      Ward, Thank You! for all you do for us.

      [WM: You don’t have Internet connectivity on this machine. Start over.]

    24. victor says:

      Ward,
      I need to add a pause into dial pattern of outbound route – asterisk@home allowed me to have 9|wwXXXXXXXXXX ‘w” for half a second pause. I need this because my trixbox connects to Nortel PBX first before going to pstn. Trixbox dial pattern config doesn’t allow me to add “w”, I tried adding it to extensions_additional.conf but it seems that it doesn’t do anything, any idieas?
      thanks for all your help

    25. Jim says:

      Ward how are you making out with Trixbox 1.2?? This is the fourth version of a@h/trix… I have installed and it is the first time I’m fairly stumped.. The server does not seem to respond to SIP registrations. IAX and ZAP came right up. So far I have not seen any information regarding this version. I now needto sniff the line to identify who the culprit is. any idea?

      [WM: Bad news here, too. 1.2 completely breaks VMware on most boxes, and it’s hit-and-miss on stand-alone machines. We’re gonna pass on this one.]

    26. Jim says:

      A network Sniff revealed the server is just not responding to the sip registration requests. Additionally, after performing an Amportal stop/start, all sip registrations came online and the zap channels went offline. If I reboot,the issue flipflops back to the original. There seems to be a process conflict. What recommendation would you make, use TB 1.1 or return to A@H 2.8 (my current starting point)?

    27. Tony Davis says:

      Ward, I have been running a@h for the last 6 months and I am now upgrading to trixbox 1.1. However, the best part of this article is your comment to the type A males. I can attest to the fact that my wife uttered those exact words only days after my first install. They eventually get used to it!!

    28. Ken says:

      Hello.
      I have installed 3 trunks, all with the same company, Unlimitel. I would like specific extensions to use specific trunks and routes so I can keep track of calling logs. How do I make sure that if one of my customers makes a call, it gets routed to their specific trunk, and proper inbound and outbound routes?
      Thanks.
      Ken.

    29. Luis Gonzalez says:

      This is one of the most complete guide for trixbox I have seen so far. Great work.

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