Home » Technology » Telephony (Page 75)
Category Archives: Telephony
The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiAsterisk PBX Management Done Right
Since the PBX in a Flash project began less than three months ago, we’ve been promising to provide Managed PBX Service and Hosted PBX Service for those that wanted these options. So today we introduce PBX-Management for PBX in a Flash, and next week we’ll bring you hosted service as well. Today’s offering really is for resellers that want to provide cradle-to-grave support for their customers although it works equally well for end-users that just want a little peace of mind.
PBX in a Flash actually grew out of the Concordiax PBX offering that has been well received in the United Kingdom. It’s founder, Joe Roper, has been using Asterisk® since the Asterisk@Home 0.3 days and was a big fan of Nerd Vittles. When he heard we wanted to develop our own Asterisk distribution with no strings attached, no surprises, and no bugs, he picked up the phone. The first release of PBX in a Flash was essentially Joe’s Concordiax PBX product minus PBX-Management. It was rock-solid reliable, stable, and easily extensible. In short, it provided all the things we were looking to bring to the open source community in an Asterisk aggregation.
So what does PBX-Management do? For openers, PBX-Management affords resellers the ability to exert some control and reduce administrative overhead on remote PBX system deployments while providing support and backups that can be managed remotely and painlessly. Each PBX checks in every 30 minutes with a heartbeat reporting on the health, condition, and IP address of the host PBX. A reseller can see the status and overall health of all installed PBX in a Flash systems in a glance.
Records are kept about every PBX in a Flash system including the number of extensions, zap hardware, uptime, IP address, database passwords, name, address, location as well as billing and IT support contact details and, most importantly, time-stamped support notes of actions that have been taken with respect to each installation. All of this information is available from a single web page.
Additionally, there are a number of actions that can be performed on any PBX in a Flash system that is subscribed to "managed care." For those of you that have remotely managed PBX systems, you know the hassles that are involved in the care and feeding of these systems. Many such systems have dynamically assigned IP addresses and clueless end-users that don’t know the difference in an IP address and a zip code.
Some PBX-Management functions still are undergoing construction and testing, but here is a brief description of all of the components:
- Lock Server – Sets the database to read only, so that your customer cannot make any changes. This was introduced for .htaccess based authentication, (maint/password) so that the customer could look at CDR, and other reports without fiddling with other settings which may break the box. With the FreePBX database authentication and ACL access on the PiaF system, this function is not as useful as it used to be.
- Suspend Server – Disables access to the database. This is as a revenue protection measure, so that if one of your customers defaults on his payment, either for support or for the initial purchase, the system will continue to work as normal, but the customer cannot access his PBX for adds and changes, but the system continues to work in the same state that it did when the PBX was suspended. We have not built in the functionality to switch off a customer PBX, as the reseller may be open to litigation if the customer loses their ability to make and receive calls, even if this is accidental. Suspend will also switch off Backups and Mail relay where used. The server can be un-suspended at any time. Once suspended, the server can then be deleted from PBX management. This is still undergoing testing.
- Move Server – Moves the PBX to another reseller account.
- Backup Server – Still under construction, but you can guess what it’s going to do.
- Clone Server – When a PBX is first installed and registered, a clone button will be available. This will allow you to clone a PBX from an existing broken system onto the replacement box, in the time that it takes for the PBX to download the backup from PBX-Management. You can only clone to a box which has zero extensions. Hence, a production system can never be overwritten with another user’s settings. As this is a function of the backup, it is not yet fully functional.
And what does PBX-Management cost? Well, for now, nothing. It’s a perfect opportunity for anyone to try it and see if it meets your needs. And until all of the functionality described above is working flawlessly, you won’t pay a dime. Once the system becomes production quality, the cost will be 25¢ per month per extension, a portion of which is returned to the PBX in a Flash development team to support future development. And, yes, you can quit at any time with no penalties of any kind… other than losing your managed care service.
How It Works. To install PBX-Management on your existing PBX in a Flash system, you first must sign up for the service at this link. Then you download a script and execute it. This is the same process used to add other components to your PBX in a Flash system. To install PBX-Management, your MySQL database and Asterisk manager passwords must not have been changed from the defaults. When you run the script, you will prompted for the PBX-Management username that you obtained when you registered for PBX-Management service. The following functions then are performed:
- Registers your system with PBX-Management.
- Downloads your logo to the main PBX in a Flash web page and to the top right hand corner of FreePBX.
- Changes the MySQL root and freePBX passwords to a random password generated by PBX-Management. The passwords are recorded in the interface.
- Changes the Asterisk Manager password and records it in PBX-Management. Customers don’t like the FreePBX reminders that default passwords still are being used.
- Sets up a cron job to run a heartbeat in /etc/pbx which reports the health, status, and IP address of the system to your PBX-Management account every 30 minutes.
Installing PBX-Management. Once you have your account established, here are the commands to execute on your PBX in a Flash system. Until testing and development is completed, Joe strongly recommends that you evaluate this on non-production systems only! We would also encourage you to make a full backup of your system before you begin. PBX in a Flash includes one of the best backup solutions in the industry so there’s no excuse for not having a good backup. Once you’re finished making your backup, log into your server as root and issue the following commands:
cd /etc/pbx
wget http://www.pbxmanagement.com/PBXScripts/ConcordiaRegister.pl
wget http://www.pbxmanagement.com/PBXScripts/ConcordiaHeartbeat.pl
chmod 711 Concordia*
./ConcordiaRegister.pl
You will be prompted for your PBX-Management username. The script is successful if it ends with the following:
Error: 000000000000000000000000000000
You then can log into PBX-Management to view your PBX on line using the credentials you used when you registered. No changes will be made to your system unless you plug in your correct login. And, as previously noted, while the system is being refined and improved with new functionality, there will be no charge to register through this link. Charges will only commence once the backup functionality is completed, and Joe will notify all customers well in advance of the cutover date providing you an opportunity to leave the program if you so desire.
How to Disable PBX-Management. If you should decide to cancel out of the PBX-Management program and wish to disable the heartbeat, first write down your MySQL root password by accessing your PBX-Management account online. Then simply remove the cron job for ConcordiaHeartbeat.pl and delete ConcordiaxRegister.pl, ConcordiaHeartbeat.pl and ConcordiaID.txt from the /etc/pbx directory on your server. Enjoy!
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Allison’s Text-to-Speech Trifecta: Cepstral, Asterisk 1.4 or 1.6, and FreePBX 2.4
If you've longed for a text-to-speech Asterisk® toolkit that sounds just like the default Allison prompts that ship with Asterisk 1.4, then today is your lucky day. We're going to walk you through installing Cepstral with Asterisk 1.4 or 1.6 and FreePBX 2.4. The icing on the cake is a new Cepstral voice that sounds just like the twin sister of Asterisk's Allison. And guess what? Just like the two Darryl's on the Bob Newhart Show, the twin sister's name is Allison, too. What a coincidence! Well, not really. Allison is actually the first TTS voice created using Cepstral's new VoiceForge™ technology. For the complete history of the development of Allison's voice for Cepstral, you can read all about it here.
Update: For the latest news on Cepstral and app_swift, visit the PBX in a Flash Forums.
Next week, in Part 2, we'll build the Nerd Vittles' Stealth AutoAttendant in FreePBX to answer your incoming calls with a separate IVR to process calls when you're not around. For those new to Nerd Vittles, the Stealth AutoAttendant answers incoming calls with a message like this: "Hi. You've reached Total Telephony Solutions. Please hold a moment while we connect your call to the next available representative." Or, for home users, the message might go something like this: "Hi. You've reached the Mundy's residence. Someone will be right with you." While the greeting message is playing, you can press keys on your phone to transfer to an extension, activate DISA, or retrieve your voicemail messages. Because the options aren't advertised in the greeting, other callers won't know they're available. We'll protect the IVR options with passwords, of course. The NoAnswer or Unavailable IVR will also include options to leave a message, transfer to your cell phone, or drop into an applications AutoAttendant. The difference in the 2008 version of these AutoAttendants is that, this time around, you can customize all of the IVR announcements and options to meet your specific needs in less than a minute. And we'll design, develop, and deploy the entire solution using FreePBX's web interface and no custom code. All of this becomes possible thanks to FreePBX and Cepstral's Allison, who will be on your payroll once we get finished up with this project.
Prerequisites. To get this working won't cost you a dime. But, once you've played with it and like it (and we know you will), you'll need to spring for the $30 to license the Cepstral Allison voice for your Asterisk system. Our advice is simple. Try it first. Then you can buy it. You'll also need a robust Asterisk 1.4 platform with Linux, Apache, SendMail, PHP, and MySQL preconfigured to support text-to-speech applications. Not that we're biased or anything, but may we recommend you give PBX in a Flash a try. You'll find complete installation instructions and the free download here.
And, during the FreePBX Training Seminar in Charleston, we'll walk you through revising the Nerd Vittles weather, news, and email text-to-speech applications to take advantage of the tremendous power that Cepstral and Cepstral's Allision now bring to your Asterisk platform. See what you're missing by not attending the FreePBX Training Seminar. Don't worry!! We'll share all of the code with you anyway, but the seminar participants will get to play with it first.
Installing Cepstral. For today, we're going to walk you through installing Cepstral with the Cepstral Allison voice. But there are numerous other voices. You can check all of them out on the Cepstral demo site. Just be sure to select only the 8kHz voices which are specifically designed to support telephony applications. Once you find the voice you like, you can decipher the download link here. Be sure you choose the correct i386-Linux version for your system. You can't use the 32-bit version on a 64-bit CentOS system, e.g. the new 64-bit ISO of PBX in a Flash 1.2. But the same license key works for both the 32-bit and 64-bit versions of the same voice. Upgrades to the 5.0 Cepstral voices are available here.
CentOS 5.x 32-bit Install. For the 32-bit version of PBX in a Flash 1.1 or 1.2, log into your system as root and enter the following commands:1
cd /root
wget http://downloads.cepstral.com/cepstral/i386-linux/↩
Cepstral_Allison-8kHz_i386-linux_5.1.0.tar.gz
tar -zxvf Cepstral*
cd Cepstral_Allison-8kHz_i386-linux_5.1.0
./install.sh
CentOS 5.1 64-bit Install. For the 64-bit version of PBX in a Flash 1.2, log into your system as root and enter the following commands:
cd /root
wget http://downloads.cepstral.com/cepstral/x86-64-linux/↩
Cepstral_Allison-8kHz_x86-64-linux_5.1.0.tar.gz
tar -zxvf Cepstral*
cd Cepstral_Allison-8kHz_x86-64-linux_5.1.0
./install.sh
After you've read the license, type yes to install the voice on your system, not -yes- as the instructions imply. Don't ask how I know. Accept the default locations for the installation. When the installation completes, issue the following command:
echo /opt/swift/lib > /etc/ld.so.conf.d/cepstral.conf
ldconfig
Now plug some speakers into your PBX in a Flash system, and type: swift "Hello World." If you want to get fancy, try this one:
swift "Hello <break time='200ms' /> World"
You can read up on Cepstral's Speech Synthesis Markup Language (SSML) here. Before we continue, you need to write down the name of the installed voice. You'll need this to register the voice later and to get Asterisk set up properly to use Cepstral. Here's the command to retrieve the voice name(s) that you've installed:
ls /opt/swift/voices
Installing app-swift. There's another important piece in getting Cepstral to play nicely with Asterisk 1.4 or 1.6, apt-swift. In the words of the author, it does four things and does them well:
* Doesn't keep the caller waiting in silence while the app generates the entire TTS output to a temp file
* Doesn't unceremoniously kill off the swift engine when done, upsetting the Cepstral license server and eating a concurrency license
* Has configurable in-memory buffering of the swift output to balance memory usage vs Swift process concurrency
* Responds to user DTMF during the speech by setting a channel variable and optionally doing a goto of the extension entered
Asterisk 1.4 Install. To install apt-swift on your PBX in a Flash/Asterisk 1.4 system:
cd /usr/src
wget http://pbxinaflash.net/source/app_swift/app_swift-1.4.2.tar.gz
tar -zxvf app_swift*
rm *.gz
cd app_swift-1.4.2
make
make install
Asterisk 1.6 Install. If you're using the newer versions of PBX in a Flash with Asterisk 1.6, you will need Darren Session's 1.6-compatible version of app-swift:
cd /usr/src
wget http://pbxinaflash.net/source/app_swift/app_swift-1.6.2.tar.gz
tar -zxvf app_swift-1.6*
rm *.gz
cd app_swift-1.6.2
make
make install
cp swift.conf.sample /etc/asterisk/swift.conf
chown asterisk:asterisk /etc/asterisk/swift.conf
Finally, you need to add a link in your search path for Cepstral and modify /etc/asterisk/swift.conf to tell it which voice you want to use with Asterisk and then restart Asterisk. Assuming you installed Allison-8kHz, here are the commands.
ln -s /opt/swift/bin/swift /usr/bin/swift
sed -i 's|David-8kHz|Allison-8kHz|' /etc/asterisk/swift.conf
amportal restart
Testing Cepstral in Your Dialplan. To be sure that everything is installed and working with Asterisk, issue this command:
asterisk -rx "core show application swift"
You should receive the following response:
-= Info about application 'Swift' =-
[Synopsis]
Speak text through Swift text-to-speech engine.
[Description]
Swift(text) Speaks the given text through the Swift TTS engine.
Returns -1 on hangup or 0 otherwise. User can exit by pressing any key.
If everything is working swimmingly, let's modify your dialplan a bit to give Cepstral a test run. Edit /etc/asterisk/extensions_custom.conf (nano -w filename) and search (Ctrl-W) for 1234. You should then see a string of code that looks something like this:
exten => 1234,1,Playback(demo-congrats)
exten => 1234,2,Hangup()
exten => h,1,Hangup()
Let's modify it so that it looks like this:
;exten => 1234,1,Playback(demo-congrats)
exten => 1234,1,Swift(Congratulations! You have installed Cepstral.)
exten => 1234,2,NoOp(Key pressed: ${SWIFT_DTMF})
exten => 1234,3,Swift(You pressed ${SWIFT_DTMF}. Goodbye.)
exten => 1234,4,Hangup()
exten => h,1,Hangup()
Save your changes (Ctrl-X, then Y, then Enter). And restart Asterisk: amportal restart. Now dial 1-2-3-4 from an extension on your PBX in a Flash system. Presto! Welcome to the World of Cepstral on your Asterisk 1.4 PBX. Should you have problems with the install, kindly post a message on the PBX in a Flash Forum. Enjoy!
Licensing Cepstral Voices. If you've made it this far with no hiccups, it's probably time to cough up your 30 bucks and make the nag messages disappear. (HINT: Read all of the comments, and you might save some money.) Keep in mind that it's $30 per simultaneous connection using Cepstral! If you're an application designer, you probably need to keep this in mind. It doesn't tie up your Cepstral voice very long to read a sentence. But reading a 7-day weather forecast is another matter. For the latter type application, it makes more sense to conserve your voice licenses by quickly generating a .wav file with Cepstral and then releasing the Cepstral engine. The same applies with IVR applications. Using Cepstral is the same PHP syntax as flite except you substitute the swift command, e.g. system("swift -f $inputfile -o $outputfile"). You then can play back the .wav file using other tools within Asterisk. Now go to this link to pay the piper. Be sure you select U.S. English language, Allison-8kHz voice, and Linux platform before you check out, or it's money down the drain. Write down the name, company (optional), and key that is issued once you fill in the blanks. Then it's back to your PBX in a Flash system as root and enter the following command. Note: it's two hyphens before the word reg-voice.
swift --reg-voice
Fill in the blanks with the information you wrote down, and you're all set. Dial 1-2-3-4 from a phone on your system again, and the nag message should be gone.
Your Name: John Q. Public
Company (if applicable): Acme Widgets
Voice: Allison-8kHz
License Key: xx-xxxxxx-xxxxxx-xxxxxx-xxxxxx-xxxxxx
Some Recent Nerd Vittles Articles of Interest...
- Join the following line and the original line with no intervening space when you encounter the ↩ character. [↩]
Build a $199 Turnkey (Green!) Asterisk 1.4 System in Less Than An Hour
It takes a lot to get our attention on the Asterisk® hardware front these days, but this one’s a genuine eyepopper in our book. For $199 (actually $223.89 including shipping and tax), WalMart delivers a brand new Everex Green gPC to your door with a 1.5GHz VIA C7-D Energy Efficient Processor, 512MB of DDR2 533MHz SDRAM, an 80GB, 7200 RPM PATA drive, DVD/CD-RW drive, 10/100 Ethernet adapter, 6 USB ports, keyboard, mouse, speakers, and one year warranty. While the machine ships with Linux preinstalled, we chose to spend 33 minutes installing the latest version of PBX in a Flash which provides CentOS 5, Asterisk 1.4, FreePBX 2.3, and the usual assortment of Linux tools: Apache, SendMail, PHP 5, phpMyAdmin, MySQL 5… and No Bugs! You can order the PC on line at this link, or you may get lucky and find one sitting in your neighborhood WalMart store. Just click on the Find in Store link on the link above. Tiger Direct and zareason.com also have the same unit, and currently it’s at the same price. For other locations, check this thread on the PBX in a Flash Forum. To say Everex can’t make these machines fast enough is really an understatement. All we can suggest is check the web sites daily. They get new shipments regularly, and they sell out just as fast. If you’re unfamiliar with the Everex brand, don’t worry. They’ve been in the PC business for decades, and their machines are built with industry-standard components. And, finally, for those of you that depend upon Asterisk systems for a living, why not build up a few of these babies as spares to have on hand. It’s the cheapest insurance you can buy!
Now that PBX in a Flash has been around for 9 whole weeks, we thought it might be helpful to walk you through the typical installation scenario that we use to bring up new systems. This gets you the latest code, all the patches, and a rock-solid system to begin your Asterisk adventure. What’s the difference in PBX in a Flash and other Asterisk 1.4 aggregations? Well, we’ll let you be the judge. Compare the Help message threads here and then here, and you’ll get the idea. So let’s get started.
Getting Started with PBX in a Flash. Just like all the other offerings, you need to begin by downloading the ISO image for PBX in a Flash (646.96 MB). As new locations for ISO downloads come on line, we will add them to the download list. Just click on the location nearest to you. Once you’ve got the image in hand, use your favorite tool to burn it to a bootable CD. Remember, your hard disk will first be erased by this install.
On the new Everex machine, insert the CD containing the pbxinaflash.iso and then reboot. After reading the initial prompts and warnings, press the Enter key to begin the installation. Or, if you want to first check the media for corruption, type linux mediacheck and then press the <Enter> key. When prompted, be sure to choose the option that erases all existing partitions and uses the default partition layout. Then choose your time zone and leave the UTC system clock option unchecked. Next choose a root password for your new system. Make it secure, and write it down. We plan to use this password for virtually everything on your new system. The install process begins. This includes MySQL, Apache, PHP, CUPS, Samba, WebMin, Subversion, SendMail, Yum, Bluetooth support, SSL, Perl, Python, the kernel development package, and much more. In about 15 minutes depending upon the speed of your PC, the install will pause to allow you to eject the CD. Click the Proceed button to continue after removing the CD. You must have an Internet connection now to complete the install so plug in a 10/100 cable if you haven’t done so already. After reboot, the system will start up with CentOS 5, then download and install Asterisk and FreePBX, and search for the necessary installation script and payload file on pbxinaflash.net. Just to repeat, If you don’t have Internet connectivity, then the installation cannot complete. When the installation finishes, reboot your system and log in as root. The IP address of your PBX in a Flash system will be displayed once you log in. If it’s blank, type service network restart after assuring that you have Internet connectivity and access to a DHCP server that hands out IP addresses. Typing ifconfig should display your IP address on the eth0 port. Write it down. We’ll need it in a minute.
Now that you’ve logged in as root, you should see the IP address displayed with the following command prompt: root@pbx:~/. If instead you see bash displayed as the command prompt and it’s not green, then the installation has not completed successfully. This is probably due to network problems but also could be caused by the time being set incorrectly on your server. You can’t compile Asterisk if the time on your computer is a date in the past! For this glitch you have to start over. If it’s a network issue, fix it and then reboot and watch for the eth0 connection to complete. Assuming it doesn’t fail the second time around, the installation will continue. Likewise, if you do not have DHCP on your network, the installation will fail because the PBX will not be given an IP address. Simply type netconfig, fill in the blanks and reboot. The install will recommence.
Required Steps to Complete the Install. There are three important things to do to complete the installation. First, from the command prompt, run genzaptelconf. This sets up your ZAP hardware as well as a timing source for conferencing. If you’re using additional hardware for your Asterisk system, we recomend removing the 56K modem when you install the cards. This will help avoid interrupt conflicts. Second, decide how to handle the IP address for your PBX in a Flash server. The default is DHCP, but you don’t want the IP address of your PBX changing. Phones and phone calls need to know how to find your PBX, and if your internal IP address changes because of DHCP, that’s a problem. You have two choices. Either set your router to always hand out the same DHCP address to your PBX in a Flash server by specifying its MAC address in the reserved IP address table of your router, or run netconfig at the command prompt and assign a permanent IP address to your server. If you experience problems with the process, see this message thread on the forum. The third configuration requirement probably accounts for more beginner problems with Asterisk systems than everything else combined. Read the next section carefully and do it now!
Getting Rid of One-Way Audio. There are some settings you’ll need to add to /etc/asterisk/sip_custom.conf if you want to have reliable, two-way communications with Asterisk: nano -w /etc/asterisk/sip_custom.conf. The entries depend upon whether your Internet connection has a fixed IP address or a DHCP address issued by your provider. In the latter case, you also need to configure your router to support Dynamic DNS (DDNS) using a service such as dyndns.org. If you have a fixed IP address, then enter settings like the following using your actual public IP address and your private IP subnet:
externip=180.12.12.12
localnet=192.168.1.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
If you have a public address that changes and you’re using DDNS, then the settings would look something like the following:
externhost=myserver.dyndns.org
localnet=192.168.0.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
Once you’ve made your entries, save the file: Ctrl-X, Y, then Enter. Reload Asterisk: amportal restart. If you assigned a permanent IP address, reboot your server: shutdown -r now.
Getting Your Machine Up to Date. Tom King, one of our lead developers, has gone to great pains to make it easy for you to always have a current system. All you have to do is type a few commands, but you do have to type them. So do it now! After logging in as root, type update-scripts to get the latest PBX in a Flash scripts installed on your system. This doesn’t run them, it merely makes them available for you to run them. Once you complete this step, you can always review the latest scripting options by typing help-pbx. Now run update-fixes to apply the latest patches to your PBX in a Flash system. When it completes, you’re up to date. If you want the latest version of Asterisk, it’s easy! Just run update-source.
Activating Email Delivery of Voicemail Messages. We’ve previously shown how to configure systems to reliably deliver email messages whenever a voicemail arrives unless your ISP happens to block downstream SMTP mail servers. Here’s the link in case you need it. As it happens, you really don’t have to use a real fully-qualified domain name to get this working. So long as the entry (such as pbx.dyndns.org) is inserted in both the /etc/hosts file and /etc/asterisk/vm_general.inc with a matching servermail entry of vm@pbx.dyndns.org (as explained in the link above), your system will reliably send emails to you whenever you get a voicemail if you configure your extensions in freePBX to support this capability. You can, of course, put in real host entries if you prefer. For 90% of the systems around the world, if you just want your server to reliably e-mail you your voicemail messages, make line 3 of /etc/hosts look like this with a tab after 127.0.0.1 and spaces between the domain names:
127.0.0.1 pbx.dyndns.org pbx.local pbx localhost.localdomain localhost
And then make line 6 of /etc/asterisk/vm_general.inc look like the following:
serveremail=voicemail@pbx.dyndns.org
Now issue the following two commands to make the changes take effect:
service network restart
amportal restart
The command "setup-mail" can be used from the Linux prompt to set the fully-qualified domain name (FQDN) of the mail that is sent out from your server. This may help mail to be delivered from the PBX. One of things mail servers do to reduce spam is to do a reverse lookup on where the mail has come from, checking that there is actually a mailserver at the other end. You can only do this if you have set up dynamic DNS or if you have pointed a hostname at your fixed IP address. Once you have done this, and assuming your ISP is cooperative, then you will receive your voicemails via email if you wish (this is set within FreePBX),and your PBX will email you when FreePBX needs an update. You set this feature in FreePBX General Settings.
If your hosting provider blocks downstream SMTP servers to reduce spam, here’s a link on the PBX in a Flash forum to get you squared away.
Setting Passwords and Other Stuff. While logged into your server as root, you can configure many of the ‘lesser’ passwords on your system (i.e. those passwords with less than root privileges) as well as phones, ZAP hardware, and other goodies. The only command you have to remember is help-pbx. Be aware that there are four different usernames and passwords that are enforced in the web interface to your PBX:
maint... to go everywhere
wwwadmin... for users needing FOP and MeetMe access
meetme... for users needing only MeetMe access
FreePBX... default username:password for admin access is admin:admin
There also is an Administration password you can set in the KennonSoft UI that displays when you point a browser to the IP address of your server. Do NOT use the same password here as it is not overly secure.
Configuring WebMin. WebMin is the Swiss Army Knife of Linux. It provides TOTAL access to your system through a web interface. Search Nerd Vittles for webmin if you want more information. Be very careful if you decide to enable it on the public Internet. You do this by opening port 9001 on your router and pointing it to the private IP address of your PBX in a Flash server. Before using WebMin, you need to set up a username and password for access. From the Linux prompt while logged in as root, type the following command where admin is the username you wish to set up and foo is the password you’ve chosen for the admininstrator account. HINT: Don’t use admin and foo as your username and password for WebMin unless you want your server trashed!
/usr/libexec/webmin/changepass.pl /etc/webmin root password
To access WebMin on your private network, go to http://192.168.0.123:9001 where 192.168.0.123 is the private IP address of your PBX in a Flash server. Then type the username and password you assigned above to gain entry. To stop WebMin: /etc/webmin/stop. To start WebMin: /etc/webmin/start. For complete documentation, go here.
Updating and Configuring FreePBX. FreePBX is installed as part of the PBX in a Flash implementation. This incredible, web-based tool provides a complete menu-driven user interface to Asterisk. The entire FreePBX project is a model of how open source development projects ought to work. And having Philippe Lindheimer’s as the Captain of the Ship is just icing on the cake. All it takes to get started with FreePBX is a few minutes of configuration, and you’ll have a functioning Asterisk PBX complete with voicemail, music on hold, call forwarding, and a powerful interactive voice response (IVR) system. There is excellent documentation for FreePBX which you should read at your earliest convenience. It will answer 99% of your questions about how to use and configure FreePBX. For the one percent that is not covered in the Guide, visit the FreePBX Forums which are frequented regularly by the FreePBX developers. Kindly post FreePBX questions on their forum rather than the PBX-in-a-Flash Forum. This helps everybody. Now let’s get started.
NOTE: PBX in a Flash comes with the IPtables firewall enabled on your system. If this causes problems with access to the FreePBX repository (for loading the FreePBX updates below), you can easily (and temporarily) turn off the firewall. Type help-pbx for assistance. Don’t forget to restart the firewall especially if your system has any Internet exposure!
Now move to a PC or Mac and, using your favorite web browser, go to the IP address you deciphered above for your new server. Be aware that FreePBX has a difficult time displaying properly with IE6 and IE7 and regularly blows up with older versions of Safari. Be safe. Use Firefox. From the PBX in a Flash Main Menu in your web browser, click on the Administration link and then click the FreePBX button. The username and password both default to admin. Click Apply Configuration Changes, Continue with Reload, and then Refresh your browser screen. Now click the Module Administration option in the left frame once FreePBX loads. Now click Check for Updates online in the upper right panel. Next, click Download All which will select every module for download and install. The important step here is to move down the list and Deselect Speed Dials and PHPAGI from the download and install options. You may also want to deselect Zork. Once these apps have been deselected, scroll to the bottom of the page and click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. Now repeat the process once more and do not deselect the two applications, then Process, Confirm, Return, Apply Config Changes, and Continue with Reload. Finally, scroll down the Modules listing until you get to the Maintenance section. Click on each of the following and choose Install: ConfigEdit, Sys Info, and phpMyAdmin. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. All three of these tools now are installed in the Maintenance section of the Tools tab of FreePBX. One final step, and you’re good to go. An update of FreePBX has been released. Click Check for Updates online. Then choose Download and Upgrade for the Core, FreePBX Framework, and System Dashboard modules. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. You now have an up-to-date version of FreePBX. You’ll need to repeat the drill every few weeks as new updates are released. This will assure that you have all of the latest and greatest software. To change your Admin password, click on the Setup tab in the left frame, then click Administrators, then Admin in the far right column, enter a new password, and click Submit Changes, Apply Configuration Changes, and Continue with reload. We’re going to be repeating this process a number of times in the next section so… when instructed to Save Your Changes, that means "click Submit Changes, Apply Configuration Changes, and Continue with reload."
Choosing Internet Telephony Hosting Providers for Your System. Before you can place calls to users outside your system or to receive incoming calls, you’ll need at least one provider (each) for your incoming phone number (DID) and incoming calls as well as a provider for your outbound calls (terminations). We have a list of some of our favorites here, and there are many, many others. Within a few weeks, we also will have some providers that will offer you some free minutes for trying their service with PBX in a Flash. You basically have two choices with most providers. You can either pay as you go or sign up for an all-you-can-eat plan. Most of the latter plans also have caps on minutes, and there are none of the latter plans for business service. In the U.S. market, the going rate for pay as you go service is about 1.5¢ per minute rounded to the tenth of a minute. The best deal on DIDs is from les.net. They charge $3.99 a month for a DID with unlimited, free incoming calls. We will have a free offer from les.net for PBX in a Flash users shortly. Another new provider with excellent service and per minute rates is Aretta Communications out of Atlanta. They also have agreed to co-host our ISO downloads for the U.S. East Coast. Thanks, Aretta! WARNING: Before you sign up for any all-you-can-eat plan, do some reading about the service providers. Some of them are real scam artists with backbilling and all sorts of unconscionable restrictions. You need to be careful. Our cardinal rule in the VoIP Wild West is never, ever entrust your entire PBX to a single hosting provider. As Forrest Gump would say, "Stuff happens!" And life’s too short to have dead telephones, even if it’s a rarity.
Setting Up FreePBX to Make Your First Call. There are four components in FreePBX that need to be configured before you can place a call or receive one from outside your PBX in a Flash system. So here’s FreePBX for Dummies in less than 50 words. You need to configure Trunks, Extensions, Outbound Routes, and Inbound Routes. Trunks are hosting provider specifications that get calls delivered to and transported from your PBX to the rest of the world. Extensions are internal numbers on your PBX that connect your PBX to telephone hardware or softphones. Inbound Routes specify what should be done with calls coming in on a Trunk. Outbound Routes specify what should be done with calls going out to a Trunk. Everything else is basically fluff.
Trunks. When you sign up with most of the better ITHP’s that support Asterisk, they will provide documentation on how to connect their service with your Asterisk system. If they have a trixbox tutorial, use that since it also used FreePBX as the web front end to Asterisk. Here’s an example from les.net. And here’s the Vitelity support page although you will need to set up an account before you can access it. We also have covered the setups for a number of providers in previous articles. Just search the Nerd Vittles site for the name of the provider you wish to use. You’ll also find many Trunk setups in the trixbox Trunk Forum. Once you find the setup for your provider, add it in FreePBX by going to Setup, Trunks, Add SIP Trunk. Our AxVoice setup (which is all entered in the Outgoing section with a label of axvoice) looks like this with a Registration String of yourusername:yourpassword@sip.axvoice.com:
allow=ulaw
authname=yourusername
canreinvite=no
context=all-incoming
defaultip=sip.axvoice.com
disallow=all
dtmfmode=inband
fromdomain=sip.axvoice.com
fromuser=yourusername
host=sip.axvoice.com
insecure=very
nat=yes
secret=yourpassword
type=friend
user=phone
username=yourusername
And our Vitelity Outbound Trunk looks like the following (labeled vitel-outbound) with no registration string:
allow=ulaw&gsm
canreinvite=no
context=from-pstn
disallow=all
fromuser=yourusername
host=outbound1.vitelity.net
secret=yourpassword
sendrpid=yes
trustrpid=yes
type=friend
username=yourusername
Extensions. Now let’s set up a couple of Extensions to get you started. A good rule of thumb for systems with less than 50 extensions is to reserve the IP addresses from 192.x.x.201 to 192.x.x.250 for your phones. Then you can create extension numbers in FreePBX to match those IP addresses. This makes it easy to identify which phone on your system goes with which IP address and makes it easy for end-users to access the phone’s GUI to add bells and whistles. To create extension 201 (don’t start with 200), click Setup, Extensions, Generic SIP Device, Submit. Then fill in the following blanks leaving the defaults in the other fields for the time being.
User Extension ... 201
Display Name ... Home
Outbound CID ... [your 10-digit phone number if you have one; otherwise, leave blank]
Emergency CID ... [your 10-digit phone number for 911 ID if you have one; otherwise, leave blank]
Device Options
secret ... 123884
dtmfmode ... rfc2833
Voicemail & Directory ... Enabled
voicemail password ... 123884
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
Now create several more extensions using the template above: 202, 203, 204, and 205 would be a good start. Keep the passwords simple, but secure! You’ll need them whenever you configure your phone instruments.
Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. We’re going to skip that tutorial today. You can search the site for lots of information on choosing providers. Assuming you have only one or two for starters, let’s just set up a default outbound route for all your calls. Using your web browser, access FreePBX on your server and click Setup, Outbound Routes. Enter a route name of Everything. Enter the dial patterns for your outbound calls. In the U.S., you’d enter something like the following:
1NXXNXXXXXX
NXXNXXXXXX
Click on the Trunk Sequence pull-down and choose your providers in the order you’d like them to be used for outbound calls.Click Submit Changes and then save your changes. Note that a second choice in trunk sequence only gets used if the calls fails to go through using your first choice. You’ll notice there’s already a 9_outside route which we don’t need. Click on it and then choose Delete Route 9_outside. Save your changes.
Inbound Routes. We’re also going to abbreviate the inbound routes tutorial just to get you going quickly today. The idea here is that you can have multiple DIDs (phone numbers) that get routed to different extensions or ring groups or departments. For today, we recommend you first build a Ring Group with all of the extension numbers you have created. Once you’ve done that, choose Inbound Routes, leave all of the settings at their default values and move to the Set Destination section and choose your Ring Group as the destination. Now click Submit and save your changes. That will set up a default incoming route for your calls. As you add bells and whistles to your system, you can move the Default Route down the list of priorities so that it only catches calls that aren’t processed with other inbound routing rules.
General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. Scroll to the bottom of the General Settings screen after selecting it from the left panel. Plug in your email address, click Submit, and save your changes. Done!
Tom King now has put together a detailed tutorial complete with screenshots to get you started with PBX in a Flash. If you are installation-challenged, have a look at the pretty pictures.
Adding Plain Old Phones. Before your new PBX will be of much use, you’re going to need something to make and receive calls, i.e. a telephone. For today, you’ve got several choices: a POTS phone, a softphone, or a SIP phone. Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack if cigs) known as a Sipura SPA-1001. It’s under $60. Be sure you specify that you want an unlocked device, meaning it doesn’t force you to use a particular service provider. Once you get it, plug the SPA-1001 into your LAN, and then plug your phone instrument into the SPA-1001. Your router will hand out a private IP address for the SPA-1001 to talk on your network. You’ll need the IP address of the SPA-1001 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The Sipura will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.
Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming your extensions were set up starting with 201, do the following. Enter House Phone as the Display Name. Enter 201 as the User ID. Enter 1234 as the Password, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it’s Registered. You’re done. Pick up the phone and dial 1234# to test it out.
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 another great SIP/IAX softphone for all platforms that’s great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up 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 (but probably not the best sound quality) is the $79 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. Some other great choices are the Aastra 9133i and the Siemans Cordless Dect phone. Do some reading before you buy. The Voxilla forums are a good place to start.
A Word About Ports. For the techies out there that want "the rest of the story" to properly configure firewalls, here’s a list of the ports available and used by PBX in a Flash:
TCP 80 - HTTP
TCP 9080 - Duplicate HTTP
TCP 22 - SSH
TCP 9022 - Duplicate SSH
TCP 9001 - WebMin
UDP 10000-20000 - RTP
UDP 5004-5082 - SIP
UDP 4569 - IAX2
UDP 2727 - Media Gateway
A2Billing Installation. Our first example of how we plan to build up PBX in a Flash systems is the installation script for A2Billing. If you want A2Billing installed on your system, then log in as root and type install-a2billing. If you don’t want A2Billing, then you don’t download or run the script. When you return to the main web page for your server after installing A2Billing, you will have two new links for A2Billing customers and A2Billing admin. You will have to install the callback funtionality manually from the docs supplied in the install. A2Billing was created by Areski, who some of you may know was responsible for Web MeetMe, ConfigEdit, and the CDR reports included with FreePBX. A2Billing is sufficiently comprehensive that it warrants an article on its own, so we will save that for another day. Some of the projects we will be looking into are how to pass calls from FreePBX to A2Billing for least-cost routing, departmental billing services, building a calling card server, and a VoIP termination platform. Then you can change your name to Ma Bell and start selling minutes to all the people for whom you have installed PBX-in-a-Flash. If you can’t wait, visit asterisk2billing.org.
Where To Go From Here. Our new script repository is now up and running at pbxinaflash.org. Tom King, the ultimate scripting guru is managing that site. So check it often. And now that PBX in a Flash 1.1 is out the door, we’ve been chomping at the bit to get all of our Nerd Vittles Goodies ported over. If you want to try them for yourself, seven are ready today at this link, and each installs in under 15 seconds: AsteriDex, Yahoo News Headlines, Weather by Airport Code, Weather by Zip Code, Worldwide Weather Forecasts, Telephone Reminders, and MailCall for Asterisk. Complete documentation for each application also is provided at the link above. And, if you still have a DBT-120 Bluetooth adapter, you’ll be happy to learn that it works out-of-the-box with PBX in a Flash on your new Everex Green PC. Dust off our recent article on Proximity Detection, and you should be in business in under 10 minutes.
For Those on the Bleeding Edge. Well, you heard it here first. As you may know, the new Asterisk 1.6 beta was released late last week on the heels of the FreePBX 2.4 beta which has been reworked to support it. Do you need to pull your hair out for days or wait months for a turnkey build? Well, no. You can download the new PBX in a Flash 1.6 Install Script today. Just don’t expect any tech support or assistance from the PBX in a Flash development team. Your dialogue on this project needs to be directly with the Asterisk 1.6 development team. What we are providing is a turnkey solution for those that are eager to experiment. Enjoy!
FreePBX Training Almost Sold Out! We’re excited about the upcoming FreePBX Training Seminar, and today we want to remind the foot-draggers that you’ve almost missed the boat. And, yes, in addition to some fantastic training and the fine cuisine of Charleston, you’re going to be treated to some once-in-a-lifetime hardware deals on the very finest Asterisk-compatible hardware cards and servers for your business. So sign up today and join the fun. This will be the hands-down very best Asterisk and FreePBX training course that money can buy.
This is a DON’T MISS opportunity to learn everything you ever wanted to know about FreePBX, Asterisk, and Linux. The course will cover IVRs, ACDs, IRQs, E911, and the rest of the alphabet as well as routing, trunking, dialplan integration, remote office configuration, echo cancellation, TDM hardware, gateways, IP phones. It’s a very full, three-day course with a half day devoted to branding and selling Asterisk systems. The seminar is being held at one of Charleston’s premier hotels, the Embassy Suites Historic Charleston, with gorgeous suites, swimming pool and exercise room, free WiFi, free breakfasts, and free cocktails every evening. There also will be evening sessions to sit down one-on-one with the FreePBX and PBX in a Flash developers with ample assistance from the quintessential Asterisk development tools: beer and whiskey!
Some Recent Nerd Vittles Articles of Interest…
100 Great Halftime Projects For You & Your Asterisk IP PBX
Over the past twelve months, we’ve covered lots of territory in building an Asterisk® PBX for home or small office use. While most of our projects have relied upon Asterisk@Home or TrixBox, many are easily accomplished using any Asterisk system running the Asterisk Management Portal (AMP) or freePBX. Today we offer our latest catalog of what’s available on the Nerd Vittles site and some great links to other Asterisk resources on the web. We’ll keep the list updated as we add new articles down the road so bookmark this new spot and check back from time to time. You can also follow our progress with a news reader. Both RSS2 and Atom feeds are available for Nerd Vittles. You’ll also find our feeds on Planet Asterisk and Planet Gadget. We try to bite off projects which can be accomplished in about 30 minutes. Your mileage may vary depending upon how many bells and whistles you choose to add to the basic project. If you have a pet project we haven’t covered, drop us a line or post a comment. Yes, we actually read ’em. Some articles include more than one project. Installing a basic TrixBox system can be completed in less than an hour. Taking into consideration that such an installation includes the full Linux operating system and complete Asterisk application plus the Apache web server, the SendMail and Asterisk mail servers, the PHP and phpMyAdmin applications, the MySQL data base management system, the SugarCRM contact management system, and the Asterisk Management Portal or freePBX, the feat is nothing short of amazing! And, if Linux isn’t your strong suit, not to worry! Check out the VMware edition of Asterisk@Home which self-installs as a Windows application on your Windows XP desktop in about 30 minutes.
With the exception of choosing a VoIP provider and telephones, all of these projects have at least one thing in common. None of them will cost you a dime. We do encourage you to support the development efforts of those that have made all of this possible by contributing even a dollar or two to the cause to help defray the costs of hardware to test all of this stuff, but that of course is left completely up to you.
Nerd Vittles User Map. Take a minute while you’re here and add yourself to our Frappr World Map compliments of Google. In making your entry, you can choose an icon: guy, gal, nerd, or geek. For those that don’t know the difference in the first two, you’re on your own. As for the last two, here’s the best definition we’ve found: "a nerd is very similar to a geek, but with more RAM and a faster modem." Our map will be a month old tomorrow and now is fairly representative of where our visitors are coming from… all 5,000 to 10,000 of you every day! If you have a favorite BBQ joint on the planet, add it to the map, too. We try to personally evaluate all BBQ recommendations! Thanks for visiting. — Ward Mundy
Introducing PBX in a Flash: The Lean, Mean Asterisk Machine (Available Now!)
Introducing U-Rang II: Windows Desktop Screenpop Utility for Asterisk
Asterisk@Home 2.7 Soup to Nuts Installation and Update Guide
Installing the VMware-edition of Asterisk@Home 2.5 on a Windows PC
Configuring Outbound Trunks for Asterisk@Home (or choose your VoIP provider below)
Mastering Outbound Call Routing and Dial Plans with Asterisk@Home
Configuring Ring Groups to Simultaneously Ring Multiple Phones
Adding Automatic Callbacks to the Asterisk Voice Mail System
Setting Up An Automatic Call Distribution (ACD) System with Asterisk
Introduction to AutoAttendants and Interactive Voice Response (IVR) Systems
Introducing the CallerID Trifecta for TrixBox 1.2.3 and freePBX 2.2.0
Handling Callers Without CallerID and Other ‘Special’ Callers
Getting Remote Dialtone with Asterisk@Home — 3 Great Solutions!
Integrating Mobile Phones and Cellphones Into Your Asterisk Dialplan
How To Make Unlimited Cellphone Calls for $5 Using Your Asterisk Dialplan
Upgrading the Asterisk Management Portal to freePBX with Asterisk@Home
The Perfect Asterisk Server: Ice Cube HU 61 (about $500) … But Any Old Clunker Will Do!
Configuring FWD for Free Outbound Toll-Free Calls with Asterisk@Home
Configuring GoIAX for Free Outbound Toll-Free Calls with Asterisk@Home
Configuring IPkall for Free Incoming Calls with Asterisk@Home
Configuring StanaPhone for Free Incoming Calls with Asterisk@Home
Securing Your Asterisk@Home System … a Must Read!
Get Your Email By Telephone: Introducing MailCall for Asterisk
Get Your News By Telephone: Introducing NewsClips for Asterisk
Introducing the CallerID Trifecta for TrixBox 1.2.3 and freePBX 2.2.0
3 Perl Ditties to Automatically Restore Names to Inbound Caller ID
TeleYapper: Asterisk Message Broadcasting System for AAH 1.5 and AAH 2.2
TeleYapper 3.0: Asterisk Message Broadcasting System for AAH 2.5 and later (we hope)
Follow-Me Phoning: Bluetooth Proximity Detection with Asterisk, Part I
Follow-Me Phoning: Bluetooth Proximity Detection with Asterisk, Part II
Follow-Me Phoning: Bluetooth Proximity Detection with Asterisk, Part III
Follow-Me Cruising: Bluetooth Proximity Detection with a TomTom GPS
Follow-Me Phoning: Bluetooth Proximity Detection with an iPhone
Follow-Me Phoning: Implementing Bluetooth Proximity Detection on a Single Asterisk Server
Free Samples. Everybody loves free samples. Not sure about a VoIP provider? Well, here’s your chance to take TelaSIP for a free test drive. Just call our Charleston number (shown in the inset) and wait for the fast busy to hang up. There’s no charge for the call because you’re never "connected." Within 15 seconds you’ll get a return call allowing you to make a FREE 10-minute phone call to almost anywhere in the U.S. All you have to do is key in the password you’re provided when you answer the return call. Keep in mind a few things. You have to call from a phone with CallerID so that the system knows where to call you back. Both legs of the call (to you and to the person you call) use GSM compression so you’re seeing TelaSIP at its most efficient but not necessarily with the best voice quality. You can set it differently on your own system if you like.
Nerd Vittles Demo Hot Line. You now can take a number of Nerd Vittles projects for a test drive… by phone! The current demos include NewsClips for Asterisk (latest news headlines in dozens of categories), MailCall for Asterisk with password 1111 (retrieve your email by phone), and Nerd Vittles Weather Forecasts by U.S. Airport Code. Just call our Stanaphone number (shown in the left margin) and take any or all of them for a spin. The sound quality may not be perfect due to performance limitations of our ancient Intel 386 demo machine. But the price is right.
Hosting Provider Special. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host up to 6 domains with 30GB of disk storage and 750GB of monthly bandwidth. Free domain registration is included for as long as you have an account. That almost doubles last month’s deal, and it really doesn’t get any better than that. Their hosting services are flawless! We oughta know. We’ve tried the best of them. If you haven’t tried a web hosting provider, there’s never been a better time. Just use our link. You get a terrific hosting service, and we get a little lunch money.
Headline News for the Busy Executive (you) and the Lazy Loafers (us). Get your Headline News the easy way: Planet Asterisk, Planet Gadget, Planet Mac, and Planet Daily. Quick read, no fluff.
Got a PDA or Web-Enabled Smartphone? Check out our new PDAweather.org site and get the latest weather updates and forecasts from the National Weather Service perfectly formatted for quick download and display on your favorite web-enabled PDA, cellphone, or Internet Tablet. And, of course, it’s all FREE!
Who Is This Guy? Ward Mundy, the author of this Asterisk article series, is a retired attorney who spent more than 30 years providing legal and technology assistance to the federal courts in the United States.
Some Recent Nerd Vittles Articles of Interest…
Some Asterisk Resolutions for the New Year
We made some New Year’s Resolutions for 2008… just as we do every year. There are the usual ones: lose weight, exercise, more quality time with the family. But you make all of those, too. This year, there are some changes in the Asterisk® landscape we’d like to see: more community participation, better training opportunities, an end to deprecating commands, and a push into major corporate and government organizations.
The Asterisk Business Model. As we count down the days to the Nerd Vittles third year birthday bash, we’ve got to say that we’ve learned a lot these past few years. The amazing part of Asterisk is really that it has survived at all. Until recently, Digium® derived almost all of its revenue off hardware sales. Fonality makes its money off hosted Asterisk solutions. Hardware vendors seem to be doing just fine as are small systems integrators. But the folks that provide the software products that make Asterisk fly are basically starving to death. The open source model has been used as a convenient way for a handful of companies to essentially profit off someone else’s work, and I’m not talking about Digium that has done much more than its fair share to contribute open source software in exchange for hardware dollars that it has earned. And this isn’t a plea for money. I retired from a cushy government job with a cushy retirement plan so starvation isn’t all that likely in my case. But, to give you an example, our recent fund-raising campaign to raise money for a dedicated server to host our forums raised a whopping $80. To those that contributed, thank you! But we have a weekly readership of roughly 50,000 people, most of whom presumably depend upon Asterisk systems every day. We’re as cheap as the next guy, but come on folks. Would $10 really change your life style that much? And we’ve heard much the same story from the FreePBX developers. So… Resolution #1 for each of you should go something like this. Find a way that you can give something back to the Asterisk community in 2008. It doesn’t have to be money! Develop an application, develop some documentation, come up with some new ideas and share them with the rest of us. But do something for somebody else without expecting something (else) for free.
The Open Source Alternative. Absent some radical shift in contributions and participation which most of us don’t expect to see, our prediction for 2008 is that the days of the open source gravy train are numbered, at least for turnkey telephony systems. Keep in mind that these are systems that most organizations used to pay hundreds of thousands of dollars to purchase and maintain. The alternative that appears to be gaining steam is to gobble up all the free software you can find and then embellish it with proprietary bells and whistles that are not made available without a charge for either the embellished product or a support contract of some type. This is a real dilemma for developers like the FreePBX and PBX in a Flash teams. As we provide more and more functionality as open source software, the "takers" gobble up the goodies, make another sale, and return almost nothing. What’s wrong with this picture? Everything!
Our Resolution #1 is to push for a review of the licensing model. We have no objection to individuals downloading and using all of our code for free forever! However, for those that profit off reselling someone else’s work product, there needs to be some type of contribution into the open source projects that comprise the bundle which is being sold by non-contributors for a handsome profit. And, no, we’re not talking about system integrators who merely charge for their time. For the most part, we’re talking about corporations that sell rebranded, open source solutions for profit. Perhaps a 5-extension license could be offered at no cost with additional extensions being sold for some fee. Another approach might be to license endpoint and/or trunk connections with vendors paying some connection fee to help defray software development costs. Nortel and others used this model for decades. These approaches, of course, also raise questions about how to divide the income between all of the open source contributors. Not sure we know the answers yet, but we’d be interested in getting your feedback and suggestions. It’s in everyone’s best interests to keep the entire Asterisk development community moving forward… and eating.
Vertical Market Penetration. Still another solution, which we happen to favor, is to license add-in code for turnkey Asterisk systems which meets the needs of specific vertical markets. For example, the hotel/motel industry could benefit immeasurably from a move to VoIP telephony. The Marriott’s, Hyatt’s, and Hilton’s of the world already have learned this. But that leaves tens of thousands of smaller hotels and motels that still are using primitive telephony systems. All it would take to make a system like PBX in a Flash a player in this market would be wakeup calls (which Nerd Vittles will provide with Telephone Reminders for Asterisk 1.4 shortly) and a method of recording from room telephones when rooms are vacant, being cleaned, ready for occupancy, or occupied. Asterisk voicemail options already run circles around the features available in most hotels. All that is missing is a way to clear voicemails when someone checks out of the hotel. And A2Billing provides hotels with an instant profit center for outbound calls worldwide. Our purpose today wasn’t to design and build a vertical market solution, but you get the idea. This wouldn’t be rocket science.
Another market which is ripe for Asterisk penetration is the medical community. Ever wondered why a full-time employee in every doctor’s and dentist’s office has to call and remind every patient of their next appointment. TeleYapper and a carefully tailored IVR would provide physicians with a far better telephony solution at considerably less cost. Tying the system into an appointments database would be icing on the cake and easy to implement since MySQL already is running on systems such as PBX in a Flash. Then there are retail stores, restaurants, department stores, WalMart’s, marinas, time shares, rental apartments, call centers, and on, and on. None of these organizations have complex telephony needs that couldn’t be met by a system like PBX in a Flash out of the box. And, with modest customization, any data processing needs could probably be met using the same system. Last but not least is the government: federal, state, and local. Do you have any idea how many separate, crappy phone systems already are in place in government offices? Many of them were installed at a cost of several thousand dollars per user. Counted up the number of government employees lately? So here’s an approach. Find a typical government organization and build them a phone system for free, except for hardware. Then get the mayor or the governor to sing its praises at the dozens of meetings these people attend every year. The sad part of this story is that we have the technical talent in the Asterisk community to produce an extremely compelling product. What’s missing is management vision coupled with a well-trained sales force to get the message across to corporations and government organizations.
Fixing the Asterisk Deal Breaker. Believe it or not, there is a serious shortcoming with Asterisk, and it has nothing to do with the feature set. It lies in the development mentality that there’s something okay about breaking application code by inventing new commands in the Asterisk Extension Language (AEL2) and deprecating (a.k.a. trashing) old ones every year or two. And now Manager 1.1 has been released in the Asterisk SVN trunk. Yikes! We’re scared to look. After the Microsoft fiasco with Visual Basic and VB .NET, one would have hoped we wouldn’t need to have this discussion.
Suppose for a moment that a handful of key commands in the C programming language were changed. The Asterisk developers would be at the front of the line screaming foul when they had to review and rewrite all of their code. Hello!! It’s the same deal when the shoe is on the other foot. This shortcoming simply has to be addressed or vertical market penetration is never going to happen. Organizations buy phone systems expecting them to work reliably for a decade or more. They also invest heavily in building customized application code to support their particular vertical market. DialPlan Functions in AEL2 dealing with timeouts, CallerID, and Asterisk Database Calls all fail if you use the Asterisk 1.2 syntax. These command language changes between Asterisk 1.2 and Asterisk 1.4 broke virtually every application ever produced for Asterisk. Furthermore, the time between versions 1.2 and 1.4 was barely a year. If you want to waste a day, try finding even a list which cross-references old Asterisk 1.2 dialplan commands to their new Asterisk 1.4 counterparts. About the best you can find is a summary of the new commands under section 6.1 here and the mishmash of old and new commands which are summarized at voip-info.org. Neither of these sites has any affiliation with asterisk.org where one would have hoped to find some information. If we’ve missed something, no doubt some fanboy will set us all straight. But, just to be clear, we’re looking for a specific link rather than an RTFM suggestion.
To put it in dollars and cents, organizations simply cannot afford to redesign and rewrite all of their application code every couple of years when someone dreams up new verbs or new ways to use parentheses, brackets, and braces. The fact that Asterisk may be free is pretty much irrelevant once the cost of rewriting all your application code is factored in. So… our plea to the core Asterisk developers is STOP DEPRECATING COMMAND SYNTAX, or you’re going to kill vertical market penetration of the product. It takes at most a few lines of code to support the 1.2 syntax of DigitTimeout(7) as well as the 1.4 syntax of Set(TIMEOUT(digit)=7). There are certainly good reasons for adding new commands to a programming language particularly to support new functionality. But why would you break every application that’s ever been written? Surely it’s not to conserve disk space or RAM in this day and age. You’re writing code for the business community, and that needs to be taken into account if Asterisk is ever going to achieve market penetration in the government and in corporate America… not to mention everywhere else! In case you couldn’t tell, if we have one pet peeve in life, it’s having to debug our own code that functioned perfectly because somebody got a bee in his bonnet to "improve" programming language syntax. <end of rant>
There’s Some Good News, Too! Well, enough of the doom and gloom. We have some terrific news to ring in the New Year as well. As most readers of this column know, FreePBX provides the pretty face for Asterisk as well as all of the smarts to get the most out of your Asterisk PBX without having to learn anything about programming. Well, so you thought! Actually, there’s an incredible number of additional things you can do with Asterisk and FreePBX once you master the FreePBX way of doing things. The problem has been that, up until now, there hasn’t been a way to get individualized training on FreePBX. Well, your prayers have been answered. The FreePBX whiz kids have put together an incredible training session, and Nerd Vittles hometown will be the host site! The FreePBX Open Telephony Training Seminar will be held in Historic Charleston, South Carolina beginning February 27 through February 29, 2008.
This is a DON’T MISS opportunity to learn everything you ever wanted to know about FreePBX, Asterisk, and Linux. The course will cover IVRs, ACDs, IRQs, E911, and the rest of the alphabet as well as routing, trunking, dialplan integration, remote office configuration, echo cancellation, TDM hardware, gateways, IP phones. It’s a very full, three-day course with a half day devoted to branding and selling Asterisk systems. The seminar is being held at one of Charleston’s premier hotels, the Embassy Suites Historic Charleston, with gorgeous suites, swimming pool and exercise room, free WiFi, free breakfasts, and free cocktails every evening. There also will be evening sessions to sit down one-on-one with the FreePBX and PBX in a Flash developers with ample assistance from the quintessential Asterisk development tools: beer and whiskey!
For those unfamiliar with Charleston, just think of it as the best of New Orleans and San Diego all rolled into one terrific Southern city known for its hospitality. By all means, bring your spouse or significant other. Charleston recently won the Reader’s Choice award as the Best Southern City. See the January 2008 issue of Southern Living magazine which is on newstands now. And, if you like New Orleans restaurants, you’ll love Charleston dining! Here’s a big hint: register early if you want to attend. Seating is limited, and the hotel will probably be full except for the rooms already blocked for this seminar. Some of your favorite vendors also will be in attendance, but we’ll save some of those surprises for the coming weeks. If you haven’t yet met Philippe Lindheimer, the lead developer of FreePBX, suffice it to say you are in for quite a treat. We also hope to have the entire PBX in a Flash development team in attendance to address your every need. So, make this New Year’s Resolution: Don’t Procrastinate or you may miss this golden opportunity. Rumor has it that, if you sign up at this link very quickly, you’ll save $600 on the registration fee! And, no, we don’t make a nickel if you attend, but if you tell ’em Uncle Ward sent you, then expect to receive a free drink at Happy Hour just like all of the other Embassy Suites’ guests. Seriously, we’re looking forward to meeting all of you. So come join us and… Happy New Year!
Some Recent Nerd Vittles Articles of Interest…
- Meet Proxmox: A Virtual Machine Platform for Asterisk
- Auto Draft
- Housekeeping 101: Managing Your Asterisk Backups and Logs
- Migrate from Incredible PBX 13-13 to Incredible PBX 2020
- DEAL OF THE DAY: OBi2182 Color WiFi Phones now $59.99
- One-Minute Wonder: It’s Incredible PBX 2021 for VirtualBox
- Taming the Condo Call Box with a Raspberry Pi & Asterisk
- SPAM Blocker & CNAM Cornucopia for Incredible PBX 2021
- Amazon’s Polly TTS Returns for Incredible PBX 2021
- Verizon 5G Service for $25/month with Unlimited Everything
- Review: The 2021 Cadillac Escalade – Everything a Tesla Isn’t
- A Fresh Look at Low-Cost VoIP Cloud Providers
- A New Day: Introducing Incredible PBX 2021 for Ubuntu 20.04
- VoIP.ms Marketplace: 5 Minutes to a Turnkey Incredible PBX
- Our Best VoIP Cloud Platform Recommendations for 2021
- Happy New Year: Introducing Incredible PBX 2021 for Debian
- Deploying an Incredible PBX 2021 PUBLIC Server
- A New Day: Introducing Incredible PBX 2021 Beta for Debian
- Capitalism 101: IBM Castrates CentOS to "Improve" RHEL
- Cyber Monday 2020: TV Deals You Can’t & Shouldn’t Refuse
- The Most Important Asterisk Component: It’s the Backups!
- Meet Acer Spin 713: A Phenomenal Computer for All Seasons
- Turbocharge Your Raspberry Pi 4 with a $45 Bootable SSD
- Groundwire for Android & iOS: The Best $10 You’ll Ever Spend
- Linphone Rocks: Free SIP Calling to Anybody, Anywhere
- Oldie But Goodie: VoIP.ms, The Most Versatile VoIP Provider
- The Ultimate Android Phone: Just $99 if you hurry
- Clearly Anywhere: The Ultimate Mobile User VoIP Companion
- SPAM Blocker & CNAM Cornucopia for Incredible PBX 2020
- Test
- Frozen in Time: Sangoma at a Crossroads or the Cliff
- Android Alert: Unmasking Your Hidden SIP Phone
- The Asterisk Superfecta: Incredible PBX 2020 Finishes First
- Amazon’s Alexa Now Available for Incredible PBX 2020
- Adding SIP URI Incoming Call Access to Incredible PBX 2020
- Finding the Perfect Laptop: Meet the System76 Lemur Pro
- Introducing OpenVPN for Incredible PBX
Just Say No: Hidden BOTs and Asterisk Don’t Mix
You may have read that a user discovered last week that current trixbox systems as recently as today include a remotely-configurable BOT, a software program that can execute certain commands locally once it receives its instructions. Reportedly, trixbox’s registry.pl "phones home" to Fonality via the Internet at 3:41 a.m. each morning to get a list of Linux commands to run. It then executes those Linux commands on your server while you’re sleeping. If the assertions of trixbox end users are true and we have no reason to believe otherwise, the existence of this remotely-configurable BOT had never been disclosed to unsuspecting users whether they were individuals or corporations. In fact, it doesn’t appear that even trixbox resellers were aware of the existence of the remotely-configurable BOT.
Let me hasten to add that Chris, Andrew, and Kerry have been good business partners of Nerd Vittles for years even though I’ve never personally met any of them. So I would never suspect that any one of them would use a tool like this for improper purposes. Our objection is more fundamental and goes to the existence of the tool itself and the failure to disclose it. Unfortunately, a remotely configurable BOT with root access privileges is a bit like giving someone a blank check… with your signature affixed. And it’s worse in this case because users had no notice that they were handing over the keys to their castle by installing and using trixbox. One can’t help wondering if Fonality management really grasps how dangerous such a system design is in this day and age. This isn’t about the commands that Fonality was executing. It’s about the commands that could be executed if this system were ever compromised. We have daily logs full of attempts to hack our systems using, you guessed it, remotely controlled BOTS.
We don’t for a minute believe that Chris Lyman and other senior management of Fonality knew about this in advance, but they certainly know now! The problem is that many programmers, in attempting to perfect the world’s finest software app, fail to consider what would happen if a tool like this one got into the wrong hands, for example the hands of a disgruntled employee. Unfortunately, just about every organization has at least one not-so-happy camper, and companies usually don’t know how dangerous such employees are until it’s too late. We obviously have no idea what safeguards Fonality may have put in place to monitor access and prevent abuse of this tool. For everyone’s sake in the Asterisk® community, we would hope LOADS OF THEM! A security breach at Fonality would basically hand over all of these trixbox systems for remote command execution as root. Or, if anyone’s DNS system is compromised, affected trixbox servers are now everyone’s worst nightmare. Hello!!!
As with many business decisions presented to organizations, the balancing act here is whether the benefits of collecting what have been represented to be marketing and usage statistics outweighed the risks if your absolute worst imaginable scenario came to pass. Merely revealing the existence of this tool made most folks shudder. And it’s still in operation. Remember, any Linux command or application could be executed with root privileges using this BOT. Take a look at the 25+ pages of comments on the trixbox forum, Google’s VoIP Users Conference, VOIPSEC, and now Slashdot if you have any doubts about the user reaction. Do we really think the crackers of the world can’t read? Is this what we want folks to remember when they hear about Asterisk?
Now imagine control of a tool like this getting into the wrong hands where someone could actually compromise the security of outside companies that knew nothing about its existence. All it took to execute commands on every newly-deployed trixbox server in the world was creation of a list of commands presumably stored on a server within the Fonality organization somewhere. Now you can appreciate how threatening a software design decision can be.
Having a hard-coded reporting mechanism that everyone is notified about up front was one thing, and that’s where this collection process began with trixbox 2. But it morphed into an open-ended, remotely-configurable BOT. And that is something quite different and downright dangerous. Suffice it to say, if we ever hope to seriously introduce Asterisk into the business community, there’s no room for BOTs in the equation, much less hidden ones. No business would knowingly tolerate an open-ended, remotely configurable BOT running on any server inside its corporate firewall, particularly one with the breadth of Linux applications at its disposal that one would normally find on trixbox systems.
This clever software should have been reviewed by senior management before it ever saw the light of day. The episode gives all of us a golden opportunity to stop and think about what we’re doing and what our fundamental obligations are to those who use our code. Hopefully, Fonality will turn this BOT off… permanently! The problem, of course, is that it’s hard to unring a bell. This BOT is already in the wild. Luckily there’s a very quick solution in this case. Here’s the command that should be added to tomorrow morning’s Fonality script: rm -f /var/adm/bin/registry.pl. We’ll all sleep better.
We hope everyone in the Asterisk development community will make a pledge to be open about the existence and scope of any future data collection processes associated with Asterisk offerings. Then users can make an informed choice on whether to use your software. A new trixbox forum member put it this way:
There is an understanding between users and developers. The understanding is often tacit but is nonetheless there. The understanding goes, "I will be executing something you wrote. I do not have the time/ability to check it all, but as professionals, I expect you to behave in a manner befitting that trust." –Minupla
We couldn’t have said it better. As for our own software, we want to be crystal clear: No Remotely-Configurable BOTs Ever! They have no legitimate purpose when weighed against the very substantial security risks they pose to all of us.
Full Disclosure. With the help of some very talented partners, Nerd Vittles now has an Asterisk-based PBX offering of its own, PBX in a Flash. It arguably "competes" against Fonality’s trixbox ce even though both offerings are free for the taking. Having written over 100 columns touting the beauties of trixbox, we felt some obligation to warn our users who may have upgraded to a more recent version of Fonality’s software. You may also want to review this article from Philippe Lindheimer, the lead developer of FreePBX.
Some Recent Nerd Vittles Articles of Interest…
Ho, Ho, Ho: Some Asterisk Stocking Stuffers from Santa
As if we haven't given you enough Christmas presents already, today we have another stocking-full of goodies that you can add to your new PBX in a Flash lean, mean Asterisk® machine. Remember the Nerd Vittles promise when we began this adventure a month ago? No Bloatware and No Bugs. We promised to provide a rock-solid, ultra-reliable Asterisk platform that could be embellished with scripts to meet your every need. We think we've already delivered the ultimate Asterisk development platform. And with over 20,000 downloads in the first month, we're pretty thrilled with the response. There seems to have been a little pent-up demand. Heh, heh! You can, of course, make up your own mind. Just visit the PBX in a Flash Download Site and join the party. It's free and completely open source so that you can add all the bells and whistles you like without begging someone else to make the changes for you. And, yes, there's loads of documentation from a step-by-step installation guide with pretty pictures to our soup-to-nuts article that'll have you up and running in about 30 minutes. And there's the new Nerd Vittles Forum with lots of tips and tricks from the whiz kids. But that's only the beginning of the fun.
The real beauty of PBX in a Flash is the ease with which you can customize it to meet your every need. And today we again throw our financial aspirations to the wind and offer you a stocking full of free add-on scripts that'll really make your nerdy pals drool. For today only, we'll call them Apps-in-a-Flash. Going forward, they're just plain old scripts. Most of these scripts will only work on PBX in a Flash systems because we want to be sure that folks using these scripts have a reliable, solid base on which to run our goodies. We also need to know how your system is configured to write the scripts. And what's so special about Apps-in-a-Flash? Well, all of these scripts install turnkey applications in under a minute flat. Of course, if you prefer a shaky platform on which to build your telephony applications, there still are plenty of other options out there for you. But, when it comes to the ultimate script site for Asterisk, we've got you covered with our all-new PBX in a Flash Script Site. Catchy name, huh? And we've got some new scripts for you today that haven't even been posted on the site yet. You may also want to visit the Best of Nerd Vittles script repository from time to time, or just sign up for the RSS Feed to stay updated automatically.
Text-to-Speech Returns! If you've been following Nerd Vittles for a while, you already know that our favorite applications for any telephony server are text-to-speech apps. The idea behind all of these applications is that you can pick up a phone to find out the same information that you could obtain with a web browser, or a television, or a radio... only faster as in instantaneous. These applications also free you from the home sofa. You can dial in for the information using almost any telephone from anywhere in the world. Well, that was the theory. For those that have endured the last year of kitchen-sink Asterisk implementations, you also know that text-to-speech was the first casualty in the migration from CentOS 4 and Asterisk 1.2 to CentOS 5 and Asterisk 1.4. Well, guess what? We've finally resolved the choppy sound glitch and text-to-speech and Flite are back with PBX in a Flash, and soon we'll have support for other text-to-speech applications as well.
Bluetooth Is Back! The other casualty in the migration to CentOS 5 and Asterisk 1.4 was Bluetooth support. That was really unfortunate because CentOS 5 has incredible Bluetooth support, and it even works with virtual applications such as VMware. And why does Bluetooth matter? Well, for long-time readers of Nerd Vittles, you'll recall that we first introduced Follow-Me Phoning several years ago. In the trade, it's known as Proximity Detection. The idea here is that, when you're in your home or office carrying your Bluetooth cellphone, your calls ring on your local phones. And, when you leave your home or office with your cellphone in hand, your Asterisk server automatically transfers your incoming calls to your cellphone. With the latest version of FreePBX which is included in all PBX in a Flash installations, you can activate Follow Me Phoning in under 5 minutes. Step 1 is to read our latest article that explained how to install everything. You can ignore the parts about needing an Asterisk 1.2 server or activating Bluetooth. We've solved all of that for you with PBX in a Flash out of the box. All you really need to do is download our Bluetooth script and configure it with your telephone extension number and the number of your cellphone. Add one line of code to your cron jobs, plug in a DBT-120 Bluetooth dongle, and it just works. The really good news is that DBT-120's used to cost $30-$40. Now there is a generic clone called Vista 2.0 which we have tested that works just as well for $1.99 plus the usual $5.99 shipping gouge. Here is the eBay link.
Welcome Home to the Asterisk Weather Station. If you've missed dialing in for a quick weather report from your Asterisk server this past year, the wait is finally over. Today, Nerd Vittles is proud to announce the return of all three of our weather applications for Asterisk: Weather by Airport Code, Weather by Zip Code, and International Weather Forecasts. There's one major difference this time around. With PBX in a Flash and our three weather scripts, you can install all three applications in less than one minute each. In fact, you'll probably spend more time typing the commands to execute the scripts than it will take the scripts to run. So let's get started. For each of these installs, simply log into your PBX in a Flash server as root and execute the commands shown below. Two words of caution: First, only run each script once on the same server. These scripts do lots of stuff behind the scenes including populating MySQL databases and modifying your Asterisk config files. So, if you run the same script more than once, you will most assuredly get a mess. Second, if you have changed your default password for MySQL, you will need to edit these scripts and insert your new password before running them.
Asterisk Weather Reports by Airport Code. After logging into your PBX in a Flash server as root, here are the commands to execute to install this application:
cd /root
wget http://bestof.nerdvittles.com/applications/weather-airport/weather.pbx
chmod +x weather.pbx
./weather.pbx
amportal restart
To use this application, pick up any phone on your PBX in a Flash system and dial 611. Then enter the three-character airport code for the weather report you wish to retrieve. Keep in mind that there are a lot more airport codes than letter combinations on your phone so we had to make some choices. You can change these if there are missing airports that you care about. Complete documentation is available on our Best of Nerd Vittles web site. Installation time: under 15 seconds.
Asterisk Weather Reports by Zip Code. After logging into your PBX in a Flash server as root, here are the commands to execute to install this application:
cd /root
wget http://bestof.nerdvittles.com/applications/weather-zip/weatherzip.pbx
chmod +x weatherzip.pbx
./weatherzip.pbx
amportal restart
To use this application, pick up any phone on your PBX in a Flash system and dial Z-I-P (947). Then enter the five-digit U.S. zip code for the weather report you wish to retrieve. No configuration of this application is really necessary. Complete documentation is available on our Best of Nerd Vittles web site. Installation time: about 30 seconds. Sorry it's so slow, but we're loading the entire zip code data base for the United States into MySQL.
International Weather Forecasts for Asterisk. After logging into your PBX in a Flash server as root, here are the commands to execute to install this application:
cd /root
wget http://bestof.nerdvittles.com/applications/weather-world/weatherworld.pbx
chmod +x weatherworld.pbx
./weatherworld.pbx
amportal restart
To use this application, pick up any phone on your PBX in a Flash system and dial 612. Then enter the one-digit code for the weather forecast you wish to retrieve. This application comes preconfigured for the ten international cities shown below. Here are the default city codes:
0. Tokyo
1. Washington
2. Berlin
3. Florence
4. Gough Island
5. London
6. Moscow
7. Sydney
8. Toronto
9. Zurich
You can easily change these to meet your needs. Complete documentation is available on our Best of Nerd Vittles web site. Installation time: under 15 seconds.
Yahoo NewsClips for Asterisk. After logging into your PBX in a Flash server as root, here are the commands to execute to install this application:
cd /root
wget http://bestof.nerdvittles.com/applications/newsclips/newsclips.pbx
chmod +x newsclips.pbx
./newsclips.pbx
amportal restart
To use this application, pick up any phone on your PBX in a Flash system and dial 511. Then enter the one-digit code for the news headlines you'd like to hear. This application comes preconfigured for nine Yahoo news feeds. These can be customized to meet your own requirements. Here are the default news feeds:
1. Top Stories
2. Sports
3. Technology
4. Showbiz
5. Business
6. Politics
7. Most Read
8. Most Sent
9. Hurricanes
Complete documentation and customization tips are available in this Nerd Vittles article. Installation time: under 15 seconds.
AsteriDex for PBX in a Flash. After logging into your PBX in a Flash server as root, here are the commands to execute to install this application. Only use this script if you live in the United States and can place outbound calls on your server by dialing 1-areacode-phonenumber. Otherwise, install the software manually using our Best of Nerd Vittles tutorial.
cd /root
wget http://bestof.nerdvittles.com/applications/asteridex4/asteridex.pbx
chmod +x asteridex.pbx
./asteridex.pbx
amportal restart
AsteriDex, as you may know, is The Poor Man's Rolodex. It has an easy-to-use web interface which you can access by pointing your web browser to the IP address of your web site. Then click the Administration tab and click on the AsteriDex button. Or you can go directly to the site: http://ipaddress/asteridex4/. You now can set the phone on your system to use for placing calls from within the web interface. If you want to access the AsteriDex directory from your cellphone and you've enabled web access through your firewall, here's the link: http://publicIPaddress/cellphone/. There's also a speed dialer which is explained in the documentation.
Complete documentation and customization tips are available in this Best of Nerd Vittles article. Installation time: under 15 seconds.
Needy Nerd's Fund. As long time readers know, we don't often solicit donations. But these are special times, and we need some help. The new Nerd Vittles Forum is already a big hit. While it uses very little bandwidth, it's a huge processor hog. We really need something other than a shared host on which to run this forum. There are a couple ways you can help. First, you can click on the Donate link at the top of this page and chip in even a little bit so that we can lease a dedicated server. Or, if you happen to be in the hosting business, you could provide a dedicated server in exchange for some terrific advertising on the Nerd Vittles site. Most of our readers have deep pockets. We're just frugal. So now's the time that you can really make a difference in this project. Thanks in advance for your help.
That's All Folks. Well, there you have it. We're going to take a breather for a few weeks. Enjoy all your new goodies. The Nerd Vittles crew wishes you a very Merry Christmas. We may slip in a few more surprises between now and New Year's, or we may not. Just check out the Nerd Vittles forum once in a while for late-breaking tips and tricks. And maybe we'll throw in another script or two as well. Enjoy!
Some Recent Nerd Vittles Articles of Interest...
- Meet Proxmox: A Virtual Machine Platform for Asterisk
- Auto Draft
- Housekeeping 101: Managing Your Asterisk Backups and Logs
- Migrate from Incredible PBX 13-13 to Incredible PBX 2020
- DEAL OF THE DAY: OBi2182 Color WiFi Phones now $59.99
- One-Minute Wonder: It's Incredible PBX 2021 for VirtualBox
- Taming the Condo Call Box with a Raspberry Pi & Asterisk
- SPAM Blocker & CNAM Cornucopia for Incredible PBX 2021
- Amazon's Polly TTS Returns for Incredible PBX 2021
- Verizon 5G Service for $25/month with Unlimited Everything
- Review: The 2021 Cadillac Escalade - Everything a Tesla Isn't
- A Fresh Look at Low-Cost VoIP Cloud Providers
- A New Day: Introducing Incredible PBX 2021 for Ubuntu 20.04
- VoIP.ms Marketplace: 5 Minutes to a Turnkey Incredible PBX
- Our Best VoIP Cloud Platform Recommendations for 2021
- Happy New Year: Introducing Incredible PBX 2021 for Debian
- Deploying an Incredible PBX 2021 PUBLIC Server
- A New Day: Introducing Incredible PBX 2021 Beta for Debian
- Capitalism 101: IBM Castrates CentOS to "Improve" RHEL
- Cyber Monday 2020: TV Deals You Can't & Shouldn't Refuse
- The Most Important Asterisk Component: It's the Backups!
- Meet Acer Spin 713: A Phenomenal Computer for All Seasons
- Turbocharge Your Raspberry Pi 4 with a $45 Bootable SSD
- Groundwire for Android & iOS: The Best $10 You'll Ever Spend
- Linphone Rocks: Free SIP Calling to Anybody, Anywhere
- Oldie But Goodie: VoIP.ms, The Most Versatile VoIP Provider
- The Ultimate Android Phone: Just $99 if you hurry
- Clearly Anywhere: The Ultimate Mobile User VoIP Companion
- SPAM Blocker & CNAM Cornucopia for Incredible PBX 2020
- Test
- Frozen in Time: Sangoma at a Crossroads or the Cliff
- Android Alert: Unmasking Your Hidden SIP Phone
- The Asterisk Superfecta: Incredible PBX 2020 Finishes First
- Amazon's Alexa Now Available for Incredible PBX 2020
- Adding SIP URI Incoming Call Access to Incredible PBX 2020
- Finding the Perfect Laptop: Meet the System76 Lemur Pro
- Introducing OpenVPN for Incredible PBX
Asterisk on Steroids: PBX in a Flash Turns 21
In honor of the 21st birthday of PBX in a Flash, we are proud to introduce version 1.1. Yep, PBX in a Flash turns 21 today. That’s 21 as in days old. Joe Roper and Tom King have put together a special birthday bundle for you. On the ISO side, we have a new version 1.1. It still relies upon CentOS 5 which is extremely stable. And, yes, we know that CentOS 5.1 has just been released. If you’ve been following along the last few weeks, then you already know that the PBX in a Flash design is different. You don’t get the kitchen sink in the ISO. Instead you get a rock-solid base operating system on which to build your Internet telephony server. When your server reboots after installation of the ISO, we then download a payload file that compiles Asterisk®, installs FreePBX, and puts you in business. For anything else you choose to add, simply download and run a script. The design theory is simple: No Bloatware and No Bugs!
Update: Much has happened in the 2+ years since this article was first published. If you’re just getting started with Asterisk and VoIP telephony, please take a look at our most recent article for the latest and greatest. Thanks for visiting. –wm
Today we’re proud to introduce the 1.1 release of PBX in a Flash for Linux, Windows, and Macs. It’s a lean, mean Asterisk machine designed to meet the needs of hobbyists as well as business users. And there’s nothing "beta" about PBX in a Flash. That explains why we’re averaging over 1,000 downloads of the ISO per day. So now we have invested 3 weeks in development and 3 weeks in production. Give it a try and see what you’ve been missing. Text-to-speech works, Bluetooth works, the platform is open, the install scripts are available for your use and modification. And PBX in a Flash is rock-solid reliable on Day 21 for a production environment.
For the early adopters, there’s always the question, "Should I upgrade?" Well, the answer is "No." If you’ve added the additional sound files and applied the two-line tweak to get rid of our one and only bug, you’ve already got a near perfect system. Here’s the link to read about any reported bugs. This version does add Asterisk 1.4.15, but the security fixes won’t affect you unless you’re using Postgres with Asterisk. And you can always upgrade Asterisk independently. Keep reading to see how easy it is! For the complete list of additions, see this message thread on the Nerd Vittles Forum.
That’s just half of the story for this birthday bash, of course. While Joe Roper has been tweaking the ISO and payload files to get them just about perfect, Tom King has been turning out some unbelievable scripts for PBX in a Flash. Take a tour of the PBX in a Flash Script Site. You won’t be disappointed. Ever wrestled with updating Asterisk on another all-in-one system? Then you know what an absolute nightmare it used to be. Not any more! Tom’s new Update Source script gets you the latest and greatest version of Asterisk any time you want it. In fact, you can download the script and use it over and over. It will always download, compile, and install the latest version of Asterisk on your PBX in a Flash system, and you’ll be back in business in about 15 minutes. And, if you’re like us, the #1 Missing Component in every other Asterisk implementation has been a way to make a mirror-image backup of your whole system that can be easily restored when there’s a system failure. There’s nothing quite like building a new PBX from scratch when your entire company has lost their phone service. Well, worry no more. Those days are behind you. For PBX in a Flash systems only, there now is a Full Backup Script that builds you a mirror image of your system in ISO format on a collection of CDs or DVDs, or on a separate USB drive, or on a Windows share, or on a remote FTP site. Your choice! And, when the dreaded day arrives, replace the failed component and boot up from the rescue disk and, presto, you’re back in business.
So, if you’re new to PBX in a Flash, keep reading. Otherwise, stop here and check back next week when we’ll begin the transition of all our previous Nerd Vittles applications to PBX in a Flash.
As mentioned above, our design model differs from other Asterisk implementations. The .iso download gives you a rock-solid CentOS 5 Linux implementation which is designed for installation on a dedicated machine. Yes, your hard disk will be erased. Once Linux is installed, the system will reboot and fetch the latest, greatest collection of add-on’s. Once the install completes, you’ll have a high-performance turnkey Asterisk PBX that’s easy to upgrade with a simple migration path to either managed PBX service or hosted PBX service. You never have to migrate if you don’t want to, and the stand-alone product will always have virtually identical functionality minus the peace of mind that comes with managed or hosted PBX service. In short, the stand-alone product isn’t ever going to be crippleware to entice you to migrate. We’ll have more about the managed and hosted options in coming weeks. There also will be a script shortly to transform your system from a solid, lean run-time system into an Asterisk development platform. In the meantime, PBX in a Flash includes Asterisk 1.4.15, FreePBX 2.3.1, Apache, MySQL, PHP, phpMyAdmin, SendMail, Perl, Flite, and much more. You also can choose from dozens of upcoming scripts to add all of the Nerd Vittles goodie bag: AsteriDex, Weather Reports, News Feeds, Email by Phone, TeleYapper, Telephone Reminders, Podcasts by Phone, and on and on. There will also be fax support, turnkey phone scripts, hosting providers with free DIDs and minutes to get you started, and lots of new stuff from developers who already are working on compatible add-on’s. You add features when you need additional functionality. Otherwise, you skip the bloatware. And you can add your own feature requests to the growing Wish List on the Nerd Vittles Forum. Visit today and sign up. Just like everything else at Nerd Vittles, it won’t cost you a dime.
For those that don’t have a dedicated Linux machine, we’ve got a VMware version of PBX in a Flash for you today as well. It’ll run on almost any fairly new Windows XP or Vista Desktop as a virtual Linux server. You can even run it as a virtual machine on a Linux desktop. Just download and install the free VMware Player. For Mac users, you’ll have to buy and install VMware Fusion. Then download and unzip pbxinaflash.zip (4K) into C:\pbxinaflash. The directory name is important, or nothing will work! Next, download the pbxinaflash.iso into C:\pbxinaflash from your favorite download site. Now open Windows Explorer and double-click on the .vmx file in your C:\pbxinaflash folder. And presto! You’ve got a PBX in a Flash… about 30 minutes to be exact. That wasn’t too hard, was it?
For those with a dedicated Linux machine, our goal today is to get everyone off on the right foot with a clean, reliable install. If you have problems with the install or spot a bug, post your questions and suggestions on the Nerd Vittles Forum. We can’t stress strongly enough that this is a collaborative adventure, and you are cordially invited to participate as not only a user but also a contributor. Whether that contribution takes the form of developing scripts or merely contributing bandwidth for .iso downloads, all of us will appreciate your efforts! Nerd Vittles hosts the payload files, the blog, and the message forum. Tom King hosts the PBX in a Flash script collection. And a group of terrific Nerd Vittles supporters now host all of the free ISO downloads. All ISO images include a checksum to make it easy for you to verify that you have The Real McCoy. At the initial installation prompt, you also can type linux mediacheck and press the <Enter> key to verify that you have a reliable CD. The size of the 1.1 ISO image is 678,385,664 bytes. And the MD5 checksum is 058a25fe23b7d1dc4265a2de2e28d8db.
Download Tips. We want to apologize for the difficulties some of you are experiencing downloading our new baby. There are LOTS of you! We are averaging over 1,000 downloads a day so, if at all possible, please use BitTorrent. It’s actually faster. Many have reported download times of under 15 minutes! In fairness to everyone, we have taken the following steps to improve the reliability of the install process. We now have six free download sites in operation with more to come. You also can obtain a free torrent as well as the VMware modules and install scripts from pbxinaflash.net and most of the other sites. If all of the free download sites are busy, we have put up a pay-as-you-go download site as well. The way this works is that you pay $5 with a credit card to download the ISO. You then can email the link to a friend who pays nothing for the download. When that friend releases the link that is provided, you can give it to another friend at no cost. Only the original download is not free, but it is dirt cheap and provides lots of backup bandwidth. So, if you get in a crunch, spring for the $5. Otherwise, check back at a different time of day and try again. Or better yet, use the torrent. The more torrents that are seeded, the easier this will be for everyone.
Getting Started with PBX in a Flash. Just like all the other offerings, you need to begin this adventure by downloading the ISO image for PBX in a Flash (646.96 MB). As new locations for ISO downloads come on line, we will add them to the download list. Just click on the location nearest to you. Once you’ve got the image in hand, use your favorite tool to burn it to a bootable CD. You’re going to need a dedicated PC for this installation unless you’re using VMware to run PBX in a Flash on your Windows or Mac desktop. A current PC in the $200-$300 price range from your favorite Big Box store should perform nicely as a dedicated Linux machine. We recommend at least 512MB of RAM and more won’t hurt. And, remember, your hard disk will first be erased by this install. You’ve been warned… twice.
On the machine you wish to use for your new PBX in a Flash system, insert the CD containing the pbxinaflash.iso and then reboot. After reading the initial prompts and warnings, press the Enter key to begin the installation. Or, if you want to first check the media for corruption, type linux mediacheck and then press the <Enter> key. For Windows users, be sure you’re doing this in the VMware window on your Desktop and not on the main screen of your PC! Erase the existing partition and accept the default for the partition layout when prompted. Then choose your time zone and leave the UTC system clock option unchecked. Next choose a root password for your new system. Make it secure, and write it down. We plan to use this password for virtually everything on your new system. The install process begins. This includes MySQL, Apache, PHP, CUPS, Samba, WebMin, Subversion, SendMail, Yum, Bluetooth support, SSL, Perl, Python, the kernel development package, and much more. In about 15 minutes depending upon the speed of your PC, the install will pause to allow you to eject the CD. Click the Proceed button to continue. After reboot, the system will start up with CentOS 5, then download and install Asterisk and FreePBX, and search for the necessary installation script and payload file on pbxinaflash.net. If you don’t have Internet connectivity, then the installation cannot complete. When the installation finishes, reboot your system and log in as root. The IP address of your PBX in a Flash system will also be displayed once you log in. Write it down, too. We’ll need it in a minute.
Now that you’ve logged in as root, you should see the IP address displayed with the following command prompt: root@pbx:~/. If instead you see bash displayed as the command prompt and it’s not green, then the installation has not completed successfully. This is probably due to network problems but also could be caused by the time being set incorrectly on your server. You can’t compile Asterisk if the time on your computer is a date in the past! For this glitch you have to start over. If it’s a network issue, fix it and then reboot and watch for the eth0 connection to complete. Assuming it doesn’t fail the second time around, the installation will continue. Likewise, if you do not have DHCP on your network, the installation will fail because the PBX will not be given an IP address. Simply type netconfig, fill in the blanks and reboot. The install will recommence. You can obtain the IP address of your server at any time by issuing the ifconfig command.
Required Steps to Complete the Install. There are three important things to do to complete the installation. First, from the command prompt, run genzaptelconf. This sets up your ZAP hardware as well as a timing source for conferencing. Second, decide how to handle the IP address for your PBX in a Flash server. The default is DHCP, but you don’t want the IP address of your PBX changing. Phones and phone calls need to know how to find your PBX, and if your internal IP address changes because of DHCP, that’s a problem. You have two choices. Either set your router to always hand out the same DHCP address to your PBX in a Flash server by specifying its MAC address in the reserved IP address table of your router, or run netconfig at the command prompt and assign a permanent IP address to your server. If you experience problems with the process, see this message thread on the forum. The third configuration requirement probably accounts for more beginner problems with Asterisk systems than everything else combined. Read the next section carefully and do it now!
Getting Rid of One-Way Audio. There are some settings you’ll need to add to /etc/asterisk/sip_custom.conf if you want to have reliable, two-way communications with Asterisk: nano -w /etc/asterisk/sip_custom.conf. The entries depend upon whether your Internet connection has a fixed IP address or a DHCP address issued by your provider. In the latter case, you also need to configure your router to support Dynamic DNS (DDNS) using a service such as dyndns.org. If you have a fixed IP address, then enter settings like the following using your actual public IP address and your private IP subnet:
externip=180.12.12.12
localnet=192.168.1.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
If you have a public address that changes and you’re using DDNS, then the settings would look something like the following:
externhost=myserver.dyndns.org
localnet=192.168.0.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
Once you’ve made your entries, save the file: Ctrl-X, Y, then Enter. Reload Asterisk: amportal restart. If you assigned a permanent IP address, reboot your server: shutdown -r now.
Activating Email Delivery of Voicemail Messages. We’ve previously shown how to configure systems to reliably deliver email messages whenever a voicemail arrives unless your ISP happens to block downstream SMTP mail servers. Here’s the link in case you need it. As it happens, you really don’t have to use a real fully-qualified domain name to get this working. So long as the entry (such as pbx.dyndns.org) is inserted in both the /etc/hosts file and /etc/asterisk/vm_general.inc with a matching servermail entry of vm@pbx.dyndns.org (as explained in the link above), your system will reliably send emails to you whenever you get a voicemail if you configure your extensions in freePBX to support this capability. You can, of course, put in real host entries if you prefer. For 90% of the systems around the world, if you just want your server to reliably e-mail you your voicemail messages, make line 3 of /etc/hosts look like this with a tab after 127.0.0.1 and spaces between the domain names:
127.0.0.1 pbx.dyndns.org pbx.local pbx localhost.localdomain localhost
And then make line 6 of /etc/asterisk/vm_general.inc look like the following:
serveremail=voicemail@pbx.dyndns.org
Now issue the following two commands to make the changes take effect:
service network restart
amportal restart
The command "setup-mail" can be used from the Linux prompt to set the fully-qualified domain name (FQDN) of the mail that is sent out from your server. This may help mail to be delivered from the PBX. One of things mail servers do to reduce spam is to do a reverse lookup on where the mail has come from, checking that there is actually a mailserver at the other end. You can only do this if you have set up dynamic DNS or if you have pointed a hostname at your fixed IP address. Once you have done this, and assuming your ISP is cooperative, then you will receive your voicemails via email if you wish (this is set within FreePBX),and your PBX will email you when FreePBX needs an update. You set this feature in FreePBX General Settings.
If your hosting provider blocks downstream SMTP servers to reduce spam, here’s a link on the PBX in a Flash forum to get you squared away.
Setting Passwords and Other Stuff. While logged into your server as root, you can configure many of the ‘lesser’ passwords on your system (i.e. those passwords with less than root privileges) as well as phones, ZAP hardware, and other goodies. The only command you have to remember is help-pbx. Be aware that there are four different usernames and passwords that are enforced in the web interface to your PBX:
maint... to go everywhere
wwwadmin... for users needing FOP and MeetMe access
meetme... for users needing only MeetMe access
FreePBX... default username:password for admin access is admin:admin
Configuring WebMin. WebMin is the Swiss Army Knife of Linux. It provides TOTAL access to your system through a web interface. Search Nerd Vittles for webmin if you want more information. Be very careful if you decide to enable it on the public Internet. You do this by opening port 9001 on your router and pointing it to the private IP address of your PBX in a Flash server. Before using WebMin, you need to set up a username and password for access. From the Linux prompt while logged in as root, type the following command where admin is the username you wish to set up and foo is the password you’ve chosen for the admininstrator account. HINT: Don’t use admin and foo as your username and password for WebMin unless you want your server trashed!
/usr/libexec/webmin/changepass.pl /etc/webmin admin foo
To access WebMin on your private network, go to http://192.168.0.123:9001 where 192.168.0.123 is the private IP address of your PBX in a Flash server. Then type the username and password you assigned above to gain entry. To stop WebMin: /etc/webmin/stop. To start WebMin: /etc/webmin/start. For complete documentation, go here.
Updating and Configuring FreePBX. FreePBX is installed as part of the PBX in a Flash implementation. This incredible, web-based tool provides a complete menu-driven user interface to Asterisk. The entire FreePBX project is a model of how open source development projects ought to work. And having Philippe Lindheimer’s as the Captain of the Ship is just icing on the cake. All it takes to get started with FreePBX is a few minutes of configuration, and you’ll have a functioning Asterisk PBX complete with voicemail, music on hold, call forwarding, and a powerful interactive voice response (IVR) system. There is excellent documentation for FreePBX which you should read at your earliest convenience. It will answer 99% of your questions about how to use and configure FreePBX. For the one percent that is not covered in the Guide, visit the FreePBX Forums which are frequented regularly by the FreePBX developers. Kindly post FreePBX questions on their forum rather than the PBX-in-a-Flash Forum. This helps everybody. Now let’s get started.
NOTE: PBX in a Flash comes with the IPtables firewall enabled on your system. If this causes problems with access to the FreePBX repository (for loading the FreePBX updates below), you can easily (and temporarily) turn off the firewall. Type help-pbx for assistance. Don’t forget to restart the firewall especially if your system has any Internet exposure!
Now move to a PC or Mac and, using your favorite web browser, go to the IP address you deciphered above for your new server. Be aware that FreePBX has a difficult time displaying properly with IE6 and IE7 and regularly blows up with older versions of Safari. Be safe. Use Firefox. From the PBX in a Flash Main Menu in your web browser, click on the Administration link and then click the FreePBX button. The username and password both default to admin. Click Apply Configuration Changes, Continue with Reload, and then Refresh your browser screen. Now click the Module Administration option in the left frame once FreePBX loads. Now click Check for Updates online in the upper right panel. Next, click Download All which will select every module for download and install. The important step here is to move down the list and Deselect Speed Dials and PHPAGI from the download and install options. You may also want to deselect Zork. Once these apps have been deselected, scroll to the bottom of the page and click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. Now repeat the process once more and do not deselect the two applications, then Process, Confirm, Return, Apply Config Changes, and Continue with Reload. Finally, scroll down the Modules listing until you get to the Maintenance section. Click on each of the following and choose Install: ConfigEdit, Sys Info, and phpMyAdmin. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. All three of these tools now are installed in the Maintenance section of the Tools tab of FreePBX. One final step, and you’re good to go. An update of FreePBX has been released. Click Check for Updates online. Then choose Download and Upgrade for the Core, FreePBX Framework, and System Dashboard modules. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. You now have an up-to-date version of FreePBX. You’ll need to repeat the drill every few weeks as new updates are released. This will assure that you have all of the latest and greatest software. To change your Admin password, click on the Setup tab in the left frame, then click Administrators, then Admin in the far right column, enter a new password, and click Submit Changes, Apply Configuration Changes, and Continue with reload. We’re going to be repeating this process a number of times in the next section so… when instructed to Save Your Changes, that means "click Submit Changes, Apply Configuration Changes, and Continue with reload."
Choosing Internet Telephony Hosting Providers for Your System. Before you can place calls to users outside your system or to receive incoming calls, you’ll need at least one provider (each) for your incoming phone number (DID) and incoming calls as well as a provider for your outbound calls (terminations). We have a list of some of our favorites here, and there are many, many others. Within a few weeks, we also will have some providers that will offer you some free minutes for trying their service with PBX in a Flash. You basically have two choices with most providers. You can either pay as you go or sign up for an all-you-can-eat plan. Most of the latter plans also have caps on minutes, and there are none of the latter plans for business service. In the U.S. market, the going rate for pay as you go service is about 1.5¢ per minute rounded to the tenth of a minute. The best deal on DIDs is from les.net. They charge $3.99 a month for a DID with unlimited, free incoming calls. We will have a free offer from les.net for PBX in a Flash users shortly. Another new provider with excellent service and per minute rates is Aretta Communications out of Atlanta. They also have agreed to co-host our ISO downloads for the U.S. East Coast. Thanks, Aretta! WARNING: Before you sign up for any all-you-can-eat plan, do some reading about the service providers. Some of them are real scam artists with backbilling and all sorts of unconscionable restrictions. You need to be careful. Our cardinal rule in the VoIP Wild West is never, ever entrust your entire PBX to a single hosting provider. As Forrest Gump would say, "Stuff happens!" And life’s too short to have dead telephones, even if it’s a rarity.
Setting Up FreePBX to Make Your First Call. There are four components in FreePBX that need to be configured before you can place a call or receive one from outside your PBX in a Flash system. So here’s FreePBX for Dummies in less than 50 words. You need to configure Trunks, Extensions, Outbound Routes, and Inbound Routes. Trunks are hosting provider specifications that get calls delivered to and transported from your PBX to the rest of the world. Extensions are internal numbers on your PBX that connect your PBX to telephone hardware or softphones. Inbound Routes specify what should be done with calls coming in on a Trunk. Outbound Routes specify what should be done with calls going out to a Trunk. Everything else is basically fluff.
Trunks. When you sign up with most of the better ITHP’s that support Asterisk, they will provide documentation on how to connect their service with your Asterisk system. If they have a trixbox tutorial, use that since it also used FreePBX as the web front end to Asterisk. Here’s an example from les.net. And here’s the Vitelity support page although you will need to set up an account before you can access it. We also have covered the setups for a number of providers in previous articles. Just search the Nerd Vittles site for the name of the provider you wish to use. You’ll also find many Trunk setups in the trixbox Trunk Forum. Once you find the setup for your provider, add it in FreePBX by going to Setup, Trunks, Add SIP Trunk. Our AxVoice setup (which is all entered in the Outgoing section with a label of axvoice) looks like this with a Registration String of yourusername:yourpassword@sip.axvoice.com:
allow=ulaw
authname=yourusername
canreinvite=no
context=all-incoming
defaultip=sip.axvoice.com
disallow=all
dtmfmode=inband
fromdomain=sip.axvoice.com
fromuser=yourusername
host=sip.axvoice.com
insecure=very
nat=yes
secret=yourpassword
type=friend
user=phone
username=yourusername
And our Vitelity Outbound Trunk looks like the following (labeled vitel-outbound) with no registration string:
allow=ulaw&gsm
canreinvite=no
context=from-pstn
disallow=all
fromuser=yourusername
host=outbound1.vitelity.net
secret=yourpassword
sendrpid=yes
trustrpid=yes
type=friend
username=yourusername
Extensions. Now let’s set up a couple of Extensions to get you started. A good rule of thumb for systems with less than 50 extensions is to reserve the IP addresses from 192.x.x.201 to 192.x.x.250 for your phones. Then you can create extension numbers in FreePBX to match those IP addresses. This makes it easy to identify which phone on your system goes with which IP address and makes it easy for end-users to access the phone’s GUI to add bells and whistles. To create extension 201 (don’t start with 200), click Setup, Extensions, Generic SIP Device, Submit. Then fill in the following blanks leaving the defaults in the other fields for the time being.
User Extension ... 201
Display Name ... Home
Outbound CID ... [your 10-digit phone number if you have one; otherwise, leave blank]
Emergency CID ... [your 10-digit phone number for 911 ID if you have one; otherwise, leave blank]
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
Now create several more extensions using the template above: 202, 203, 204, and 205 would be a good start. Keep the passwords simple. You’ll need them whenever you configure your phone instruments.
Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. We’re going to skip that tutorial today. You can search the site for lots of information on choosing providers. Assuming you have only one or two for starters, let’s just set up a default outbound route for all your calls. Using your web browser, access FreePBX on your server and click Setup, Outbound Routes. Enter a route name of Everything. Enter the dial patterns for your outbound calls. In the U.S., you’d enter something like the following:
1NXXNXXXXXX
NXXNXXXXXX
Click on the Trunk Sequence pull-down and choose your providers in the order you’d like them to be used for outbound calls.Click Submit Changes and then save your changes. Note that a second choice in trunk sequence only gets used if the calls fails to go through using your first choice. You’ll notice there’s already a 9_outside route which we don’t need. Click on it and then choose Delete Route 9_outside. Save your changes.
Inbound Routes. We’re also going to abbreviate the inbound routes tutorial just to get you going quickly today. The idea here is that you can have multiple DIDs (phone numbers) that get routed to different extensions or ring groups or departments. For today, we recommend you first build a Ring Group with all of the extension numbers you have created. Once you’ve done that, choose Inbound Routes, leave all of the settings at their default values and move to the Set Destination section and choose your Ring Group as the destination. Now click Submit and save your changes. That will set up a default incoming route for your calls. As you add bells and whistles to your system, you can move the Default Route down the list of priorities so that it only catches calls that aren’t processed with other inbound routing rules.
General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. Scroll to the bottom of the General Settings screen after selecting it from the left panel. Plug in your email address, click Submit, and save your changes. Done!
Tom King now has put together a detailed tutorial complete with screenshots to get you started with PBX in a Flash. If you are installation-challenged, have a look at the pretty pictures.
Adding Plain Old Phones. Before your new PBX will be of much use, you’re going to need something to make and receive calls, i.e. a telephone. For today, you’ve got several choices: a POTS phone, a softphone, or a SIP phone. Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack if cigs) known as a Sipura SPA-1001. It’s under $60. Be sure you specify that you want an unlocked device, meaning it doesn’t force you to use a particular service provider. Once you get it, plug the SPA-1001 into your LAN, and then plug your phone instrument into the SPA-1001. Your router will hand out a private IP address for the SPA-1001 to talk on your network. You’ll need the IP address of the SPA-1001 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The Sipura will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.
Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming your extensions were set up starting with 201, do the following. Enter House Phone as the Display Name. Enter 201 as the User ID. Enter 1234 as the Password, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it’s Registered. You’re done. Pick up the phone and dial 1234# to test it out.
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 another great SIP/IAX softphone for all platforms that’s great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up 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 (but probably not the best sound quality) is the $79 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. Some other great choices are the Aastra 9133i and the Siemans Cordless Dect phone. Do some reading before you buy. The Voxilla forums are a good place to start.
A Word About Ports. For the techies out there that want "the rest of the story" to properly configure firewalls, here’s a list of the ports available and used by PBX in a Flash:
TCP 80 - HTTP
TCP 9080 - Duplicate HTTP
TCP 22 - SSH
TCP 9022 - Duplicate SSH
TCP 9001 - WebMin
UDP 10000-20000 - RTP
UDP 5004-5082 - SIP
UDP 4569 - IAX2
UDP 2727 - Media Gateway
A2Billing Installation. Our first example of how we plan to build up PBX in a Flash systems is the installation script for A2Billing. If you want A2Billing installed on your system, then log in as root and type install-a2billing. If you don’t want A2Billing, then you don’t download or run the script. When you return to the main web page for your server after installing A2Billing, you will have two new links for A2Billing customers and A2Billing admin. You will have to install the callback funtionality manually from the docs supplied in the install. A2Billing was created by Areski, who some of you may know was responsible for Web MeetMe, ConfigEdit, and the CDR reports included with FreePBX. A2Billing is sufficiently comprehensive that it warrants an article on its own, so we will save that for another day. Some of the projects we will be looking into are how to pass calls from FreePBX to A2Billing for least-cost routing, departmental billing services, building a calling card server, and a VoIP termination platform. Then you can change your name to Ma Bell and start selling minutes to all the people for whom you have installed PBX-in-a-Flash. If you can’t wait, visit asterisk2billing.org.
Where To Go From Here. Our new script repository is now up and running at pbxinaflash.org. Tom King, the ultimate scripting guru is managing that site. So check it often. And now that PBX in a Flash 1.1 is out the door, we’re chomping at the bit to get all of the Nerd Vittles Goodies ported over. If you want to try one yourself, start with Weather by Airport Code which installs in under 5 minutes. Put your dialplan code in extensions_custom.conf, and be sure to heed the Asterisk 1.4 mod which is explained in the article. Hold off on AsteriDex because we have an updated version of that one for you within the next few weeks. If you still have a DBT-120 Bluetooth adapter, you’ll be happy to learn that it works out-of-the-box with both the Linux and Windows versions of PBX in a Flash. Dust off our two-year old articles on Proximity Detection and see if you can’t configure your system to transfer your phone calls to your cellphone when you leave the house carrying your Bluetooth cellphone. If not, we’ll walk you through it with an iPhone in coming weeks. Now spend some time reading up on FreePBX and enjoy your new toy. Santa came early this year!
Bug Fixes. We’re back down to zero bugs with PBX-in-a-Flash 1.1, but we wanted to provide a way to report them and to address them. So there’s a new PBX-in-a-Flash forum dedicated to bug reporting and fixes. Be sure to visit once in a while just to keep your system current.
Some Recent Nerd Vittles Articles of Interest…
Free Asterisk Solutions