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The Incredible PBX: Remote Phone Meets the Travelin’ Man

Monday, June 28, 2010 / 18 Comments on The Incredible PBX: Remote Phone Meets the Travelin’ Man

Ever wrestled with one of those thorny problems for weeks only to wake up in the middle of the night with the answer? Thus was born Travelin’ Man, a web- based, one-click Asterisk® application that automatically reconfigures your Asterisk PBX to enable remote SIP phone access from your cellphone, iPad, remote PC, NetBook, or desktop telephone.

News Flash: Be sure to read our latest article introducing Travelin’ Man 3, a completely new security methodology based upon FQDN Whitelists and DDNS. In a nutshell, you get set-it-and-forget-it convenience and rock-solid VoIP security for your Cloud-based PBX or any PBX in a Flash server that’s lacking a hardware-based firewall and you get both transparent connectivity and security for your mobile or remote workforce.

If you’ve read the Incredible PBX series of articles on Nerd Vittles, you already know what a thorny problem remote phone access is if you want to preserve the overall security of your server. Indeed, our recommendation has been to leave SIP access closed on your hardware-based firewall because of the dangers inherent in activating remote SIP access. Now we have a better idea!

Today’s new approach works like this. First, we’ll run a little script that secures all of your extensions with permit entries locking down all these connections to the IP address range within your private network. Then we’ll open the SIP and RTP ports on your hardware and software firewalls and map these ports to your Asterisk server’s private IP address. With this setup, no one can attempt remote SIP logins to your server because Asterisk blocks all SIP extension connection attempts except those originating inside your LAN. To manage external phone connections to your server, the install script creates a new virtual Apache web server on your Incredible PBX using port 83. We’ll enable and map TCP port 83 on your hardware and software firewalls to your server as well. Web access with port 83 is limited to running the Travelin’ Man app to activate external phones.

Now we’re ready to set up access to your server for remote devices. For each extension you wish to enable for remote access, we’ll create a special web directory using an obscure, random file name which will serve as the web link for the Travelin’ Man web app. For example, in the diagram above, directory 184778 manages extension 501, directory 2389957h manages extension 701, and directory 6993h5j manages extension 702. This is accomplished by simply changing the extension number in the index.php script stored in each directory.

When one of these web links is accessed remotely, the PHP script will automatically reconfigure Asterisk to enable access to the designated SIP extension on your server using the remote IP address from which the web page was accessed. And, of course, there’s an additional layer of SIP security as well. You still need your extension credentials to actually log in to your server with a softphone to place and receive calls. The Travelin’ Man installation process takes only a couple minutes, and the remote SIP activation procedure takes just a couple seconds each time you want remote access from a different location. Here’s a quick example of how it actually works.

Let’s assume we want to use the new $3.95 Bria SIP softphone on an iPad to connect as extension 501 on our Incredible PBX back at home. The problem is that the dynamic IP address of your iPad changes at each new site on your itinerary. Some locations have WiFi while others only have 3G connections.

First, we’ll generate an icon to run Travelin’ Man from your iPad desktop. Use the same procedure with an iPhone or iPod Touch, and there’s a similar procedure for Android devices.¹ You only have to do this once. Start up Safari on the iPad to access the new port 83 web server at the random web address the installer created to support extension 501. That web address is something like this using your own FQDN²: http://myserver.dyndns.org:83/184778. After establishing the link once, we’ll hit the + button in Safari and choose Add to Home Screen. This creates the TravelMan icon on the iPad. See the screenshot below of our demo iPad setup which used extension 221 instead of 501.

Once configured, it’s just two clicks to enable your remote phone anywhere: click once on the TravelMan icon. When your IP address is confirmed, return to your Home Screen and click the Bria softphone icon to establish a SIP connection back to your server. Behind the scenes, the Travelin’ Man application will generate the required permit entry for your remote IP address mapping it to the designated extension on your server, and then it will reload your SIP settings to make your Asterisk server accessible to the Bria softphone in your hotel room. The entire process takes only a couple seconds.

If your company happens to have a dozen traveling salesmen, then you’d simply assign a dedicated extension to each employee and create secure directory names for each person (e.g. 2389957h and 6993h5j in diagram above) with a copy of the Travelin’ Man app configured for that employee’s extension number. Now your entire mobile workforce has connectivity back to the home office from any location on the globe. And, when an employee leaves the company and another arrives, just create a new name for the old employee’s web directory to preserve the security of your system (e.g. 184778 in our example becomes 78hd773). Keep in mind that each time the Travelin’ Man app is run for any extension, it wipes out any previously authorized IP address entry for that extension. Thus, the security of your Incredible PBX is always preserved.

Prerequisites. Before proceeding with today’s install, you must be running a stock install of Incredible PBX with PBX in a Flash behind a properly-secured, hardware-based firewall³. We recommend the latest version of Asterisk 1.4 because it addresses a SIP vulnerability that might cause you problems if malformed SIP packets are targeted at your server. The current release of PBX in a Flash (1.7.5.5 Silver) is ideal, but any version of PBX in a Flash can be brought current with Asterisk using the update-source and update-fixes tools. Travelin’ Man assumes that you have the Incredible PBX base install of extensions: 501 plus 701-715. You can obviously add more or remove some, but you’ll need to manually adjust sip_custom_post.conf to reflect your actual extension list after the install completes.

The installer has been encrypted for your/our own protection. In source form, the script would allow anyone to defeat the Incredible PBX requirement. Doing so would mean the required IPtables security component would not be in place and properly configured to protect the underlying system from attack. So we’ve opted to play Big Brother to avoid potential security problems for all of us down the road. This article clearly explains all the necessary components if some folks want to roll their own version. We just don’t want the responsibility if something goes horribly wrong. As Forrest Gump would say, "Shit Happens." 🙂 If you don’t believe it, check out the latest security scramble in the trixbox forums.

Installation. Now we’re ready to get started. So log into your Incredible PBX as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/travelinman.tar.gz
tar zxvf travelinman.tar.gz
./travelinman.x

NOTE: If you’re using PIAF2 with CentOS 6.2, you’ll need to use the updated version of Travelin’ Man because of a syntax change in the Apache config file:

cd /root
wget http://incrediblepbx.com/travelinman2.tar.gz
tar zxvf travelinman2.tar.gz
./travelinman2

The first step in the install procedure is to lock down access to all of your extensions to your private LAN subnet. In case you ever want to do this on another server not running the Incredible PBX, here’s a link to our privip.sh shell script that shows how to do it. This should work on most FreePBX-based Asterisk systems.

Once the extensions are locked down, the script will modify your IPtables and Apache configurations to permit web access on port 83. Next, it will adjust your Asterisk setup to support the Travelin’ Man permit scheme. This involves reworking of sip_custom_post.conf so that permit settings for individual extensions can be stored in files named 501.inc, 701.inc, etc. Finally, the installation procedure will set up a single web site to support extension 501 with a randomized directory name for remote access.⁴ This setup will be stored in /var/www/travelman. To activate support for additional extensions, you would simply copy the subdirectory giving it a new random name: cp -r dir1 dir2. Then edit config.php in the new subdirectory and change the $extension entry.

To complete the install, you must reconfigure your hardware-based firewall and map the following ports to the private IP address of your server:

TCP 83
UDP 5060
UDP 10000-20000

When the installation is completed, it will show you how to access the new web site for extension 501 using either a fully-qualified domain name or a public or private IP address. Now just follow the steps at the beginning of this article to set up your Android or iDevice, and test things out. Enjoy!

Reminders: Be sure to review the comments to this article and the related support forum thread for a week or two for late-breaking enhancements and issues. Also, Incredible PBX comes preconfigured with call forwarding activated for extension 501. Don’t forget to either disable it or set up a real call forwarding number for extension 501 if you want your cellphone to ring. From any extension on your server, just dial *72501 to set up call forwarding. To cancel call forwarding and pass calls directly to the registered 501 softphone, dial *74 and enter 501. Also be aware that the default RingAll ring group (700) configuration on Incredible PBX systems does not include extension 501. So add 501 if you want your remote extension to ring for incoming calls.

The Incredible PBX: Basic Installation Guide

Adding Skype to The Incredible PBX

Adding Incredible Backup… and Restore to The Incredible PBX

Adding Multiple Google Voice Trunks to The Incredible PBX

Adding Remotes, Preserving Security with Incredible PBX

Continue reading Basic Installation Guide, Part II.

Continue reading Basic Installation Guide, Part III.

Continue reading Basic Installation Guide, Part IV.

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.

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…

To create a desktop icon for Travelin’ Man on Android devices, navigate to the link with your browser. Then save the link as a Bookmark by clicking the Star icon in your browser then click Add. Return to the Home Screen and, from the screen on which you wish to add the icon, touch and hold your finger on the screen. When the Add to Home Screen menu appears, choose Shortcuts then Bookmarks and select the link you previously saved. As with iDevices, you only have to do this once. [↩]
FQDN = Fully-qualified domain name [↩]
We recommend the dLink Router/Firewall. Low Cost: $35 WBR-2310 Best: DGL-4500 [↩]
If you’d like to download the web site code independently from the Travelin’ Man install procedure, here’s the link. [↩]

The Incredible PBX: Adding Remotes, Preserving Security

Monday, June 7, 2010 / 16 Comments on The Incredible PBX: Adding Remotes, Preserving Security

Unlike most Asterisk®-based PBXs which are insecure as installed and leave it to you to implement sufficient safeguards to preserve the integrity of your system, the Incredible PBX is delivered with rock-solid, air-tight security already in place. Because it is designed to operate behind a hardware- based firewall, what you'll be doing when you want to add functionality with the Incredible PBX is loosening security rather than tightening it. The trick, of course, is to do it in a way that doesn't compromise the overall integrity of your system. As delivered, the Incredible PBX relies upon four layers of network security: a hardware-based firewall of your choice¹, a preconfigured IPtables software-based Linux firewall, preconfigured Fail2Ban to monitor your logs for suspicious activity and to block specific IP addresses when abuse is detected, and random passwords for all extensions and DISA connections.

If you installed the Incredible PBX using SIPgate as the intermediate provider with Google Voice, then your hardware-based firewall should have no ports opened and forwarded to your server. If you used IPkall, then only UDP 4569 has been opened and forwarded to your server. And the Incredible PBX IPtables setup for IAX restricts access to just a few IP addresses to support IPkall.

There are obviously situations in which you will want or need additional connectivity. The most likely one involves activation of SIP telephones at remote locations, such as a branch office, or Grandma's house or a relative in college. The other obvious use is with cellphones and PDAs that support SIP clients such as Android phones, iPhones, and iPads.²

What we'd recommend you not do is open the SIP floodgate to your PBX by providing unrestricted inbound SIP access, but we'll show you how if you really want or need this functionality. As desirable as this can be, it is accompanied by an array of security issues that really are not worth the risks unless you know what you're doing and you're willing to stay on top of security updates and keep your system patched.

Let's first tackle how to provide limited inbound SIP functionality without selling the farm. If the remote site has a fixed IP address, the procedure to allow remote access to your server is fairly straight-forward: just map the SIP ports on the hardware-based firewall to your server (UDP 5000:5082 and UDP 10000:20000) and then restrict SIP access using IPtables to the remote IP address as well as the subnet of your private LAN. You can decipher your private subnet by running status. If your server's IP address is 192.168.0.123, then your private subnet would be 192.168.0.0. The IPtables firewall settings are stored in /etc/sysconfig/iptables. Edit that file and find the line that looks like this:

-A INPUT -p udp -m udp --dport 5000:5082 -j ACCEPT

Delete or comment out this entry with a leading # and insert new entries that look like the following using the public IP address(es) you wish to add plus the private subnet:

-A INPUT -p udp -m udp -s 141.146.20.10 --dport 5000:5082 -j ACCEPT
-A INPUT -p udp -m udp -s 141.146.20.11 --dport 5000:5082 -j ACCEPT
-A INPUT -p udp -m udp -s 192.168.0.0/255.255.0.0 --dport 5000:5082 -j ACCEPT

After making the changes, save the file: Ctrl-X, Y, then Enter. Then restart IPtables: service iptables restart.

Unfortunately, in many situations, the remote phone or cellphone uses an Internet connection with a dynamic IP address. So we don't know the actual IP address that will be assigned. There are a number of solutions to this problem, and we'll rank them in our order of preference. First, spend the $200 and install another Incredible PBX at the remote site. Then the two servers can be linked with IAX connections between the servers making connectivity between the systems totally transparent. Second, install VPN routers at both sites and use a private IP address to establish connectivity with the host system. In this situation, you will have the equivalent of a fixed IP address for the remote device which makes it the equivalent of the fixed IP address solution above. Third, install OpenVPN on your host system and purchase a SIP phone or cellphone that supports VPN connectivity. Most of the high-end SNOM SIP phones have this functionality as do Android phones, iPhones, and iPads. With this setup you also have the equivalent of a fixed IP address, even though it's on a virtual private network. Fourth, talk to the Internet service provider at your remote site and obtain the range of IP addresses that DHCP hands out to those using their services... or just make an educated guess.³

BEFORE Activating Full SIP Connectivity. OK. We hear you. You travel for a living, and the IP address of your cellphone changes hourly, all day, every day of the year. Then, yes, you are a candidate for a full-fledged Asterisk server with unlimited SIP access. Before covering how, let's review what responsibilities go with running such a server. Bear in mind that one compromised SIP password or otherwise vulnerable application on your server (including Asterisk, FreePBX, SSH, and hundreds of others), and you may very well be the proud owner of a whopping phone bill. And we're not talking hundreds of dollars. It could very well be tens of thousands of dollars. And it doesn't take weeks or months. It could be a few hours.

Baker's Dozen SIP Security Checklist

1. Keep Asterisk Current & Patched
2. Keep FreePBX Current & Patched
3. Make Frequent Backups
4. Visit PBX in a Flash Forums Regularly
5. Subscribe to PBX in a Flash RSS Feed
6. Secure Alphanumeric Extension Passwords
7. Secure DISA, VMail, Root, FreePBX Passwords
8. Lock Down Extensions with Deny/Permit
9. Turn Off Recurring Payments with Providers
10. Restrict Trunks to 1-2 Simultaneous Calls
11. Tighten Dialplan by Removing Wildcards
12. Eliminate Intl & Toll Calls With Providers
13. Check FreePBX Call Logs Daily for Abuse

Baker's Dozen SIP Security Checklist. Before opening the floodgates, let's review what you need to do. First, you'll need to run the very latest version of Asterisk... all the time. This means you need to monitor asterisk.org, and keep your system up to date by running update-scripts, update-source, and update-fixes regularly. The default version of Asterisk on current PBX in a Flash and Incredible PBX builds is extremely reliable, but it contains SIP and IAX vulnerabilities which should not be exposed directly to the Internet! Second, you need to run the latest version of FreePBX and apply all patches as they are released. Third, you need to make frequent backups appreciating that sometimes the Asterisk and FreePBX developers get things horribly wrong, and stuff that used to work no longer does. Believe it or not, they're human! Fourth, you need to visit the PBX in a Flash Forums daily and keep abreast of security alerts and bug reports on CentOS, Asterisk, and FreePBX. Fifth, you need to subscribe to the PBX in a Flash RSS Feed which provides regular security alerts when there are reported problems. Sixth, you need to really secure your extension passwords with very long, complex alphanumeric passwords. Ditto for your root and FreePBX passwords! Seventh, for DISA and voicemail, these passwords need to be numeric, complex, and extra long. Eighth, you need to lock down as many of your extensions as possible with deny/permit settings to restrict the IP addresses of those extensions. If you only have one or two remote SIP extensions with dynamic IP addresses, then all of the rest should have deny/permit entries! Ninth, turn off recurring payments with all of your telephony providers and keep minimal funds available in all of your accounts. This means you'll have to monitor these accounts to make sure they are not deactivated for lack of funds. Tenth, restrict all of your trunks to one or at most two simultaneous calls to reduce your call exposure in the event someone breaks into your system. Eleventh, tighten up your Trunk Dial Rules and eliminate any entries that would permit calls to anywhere in the world! If you don't regularly make international calls, there's absolutely no reason to have such entries in your dialplan. If you still have Ma Bell PSTN lines, this is even more important. In fact, consider eliminating long distance access to all of these trunks. Twelfth, where possible, configure your provider accounts to eliminate international and toll calls of all varieties. Finally, check your FreePBX call log every day to make certain no one is making calls on your nickel.

If you are unwilling or unable to perform these Baker's Dozen steps while continuing to monitor the sites provided and recheck your setup regularly (at least every week), don't activate unrestricted SIP access to your server.

Other Options. Consider using an intermediate provider such as voip.ms to provide SIP URI access to your server. Keep in mind that having a registered connection between your server and a VoIP provider alleviates the need to punch a hole in your firewall. So the idea here is to sign up for an inexpensive voip.ms account and set up the trunk connection with your server as either an IAX or SIP account with an always-on connection. Then voip.ms gives you the option of activating a SIP URI as part of a subaccount setup. Just create an internal extension on their server, and this will generate a SIP URI, e.g. 123456666@sip.us4.voip.ms where 12345 is your voip.ms account number and 6666 is the internal extension you created. This lets you connect directly with your server through the SIP URI from anywhere once you map this subaccount to an extension or IVR on your server. The charge for SIP URI calls is only $.001 per minute. The last step is to use this SIP URI in your remote SIP phone to connect back to your server. You can take advantage of the full range of Asterisk functions once these calls reach your server including IVRs and DISA. The approach is not only simple to implement, but it's also safe and economical.

There are some other alternatives as well. Use something like Google Voice or Ooma to redirect calls to your cellphone when you're traveling. Or buy an Ooma for Grandma or a MagicJack for Joe College. These options also are safe, secure, and quite inexpensive.

Just Released: Remote Phone Meets Travelin' Man

Activating Inbound SIP on Your Server. If you still are hell-bent on opening SIP access to your server, the Incredible PBX already is preconfigured to support it. Just map the SIP ports on your hardware- based firewall to your server (UDP 5000:5082 and UDP 10000:20000). Once activated, anyone can reach you through the following SIP URI using the actual public IP address of your server: mothership@12.34.56.78. You also can adjust the e164 trunk in FreePBX to route inbound calls to any destination desired. Then register your phone number on e164.org and others can call you at no cost using your traditional phone number. Enjoy!

The Incredible PBX: Basic Installation Guide

Adding Skype to The Incredible PBX

Adding Incredible Backup... and Restore to The Incredible PBX

Adding Multiple Google Voice Trunks to The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Basic Installation Guide, Part II.

Continue reading Basic Installation Guide, Part III.

Continue reading Basic Installation Guide, Part IV.

Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won't have to wait long for an answer to your questions.

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.

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...

We, of course, continue to recommend a dLink Router/Firewall. Low Cost: $35 WBR-2310 Better: DIR-825 Best: DGL-4500 [↩]
We recommend the free SipAgent client for Android devices and the commercial Acrobits Softphone for iPods and iPads. [↩]
Adding an entry like the following would dramatically reduce the likelihood of a SIP attack: -A INPUT -p udp -m udp -s 141.146.0.0/255.255.0.0 --dport 5000:5082 -j ACCEPT [↩]

Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

Monday, March 15, 2010 / 18 Comments on Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

It’s been an exciting couple of weeks watching the overwhelmingly positive response to our release of Orgasmatron 5.1. With this version, we introduced a new Asterisk® security model that took into account the ever-increasing security risks posed by exposing web and telephony servers to direct Internet access. The bottom line is this. If your telecom requirements still can be accomplished by placing a server securely behind a $35 hardware-based Internet firewall with no Internet exposure, then it makes absolutely no sense to dangle such a tempting target in front of the world’s most nefarious creeps.

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Our experience suggests that the only trade off with this new approach is the inability to receive anonymous SIP calls… a small price to pay considering the potential financial and computer risks involved. You still can place outbound VoIP calls as well as placing and receiving calls using any of the phone numbers registered on your new PBX in a Flash server. And, thanks to Google Voice, SIPgate, and IPkall, all inbound calls are free, and all outbound calls to numbers in the U.S. and Canada are free as well.

If a SIP URI and your own Freenum/ISN number are simply features you can’t live without, sign up for a voip.ms IAX account, and you’ll get a SIP URI for free. Inbound SIP URI and Freenum/ISN calls will set you back $1 for every 1,000 minutes billed in 6 second increments.

Or you can sign up for a free IP Freedom CallCentric account and configure a new SIP trunk in FreePBX by following these directions. Once configured, your new server SIP URI will be 1777xxxxxxx@in.callcentric.com where xxxxxxx is your assigned 7-digit CallCentric number.

Keep in mind that a new security vulnerability has been found with either Asterisk or FreePBX almost monthly. The chart below tells you why. With virtually limitless attack surfaces because of the number of interrelated components in CentOS, Asterisk, and FreePBX comes enormous and recurring potential for remote compromise of these systems. Rather than play this cat-and-mouse security game with the underworld, the Orgasmatron design changes the paradigm. It lets you use any (secure or insecure) version of Asterisk and FreePBX without worrying about any outside attacks. Do passwords on your new server matter? Not really… unless there is someone inside your firewall that you don’t trust. 🙄 Are we going to secure them anyway? Absolutely. But instead of the constant worry over new security vulnerabilities, Orgasmatron 5.2 lets you enjoy exploring the world of Asterisk and VoIP telephony with an incredibly rich feature set that you won’t find anywhere else, period! We’ll resist making any other device analogies, but the idea here is to protect the good guy (you!) while keeping the bad guys out. No penetration. No worries. Simple as that.

In our former life working for a living, we actually procured and managed multimillion dollar PBXs as part of our "other duties as assigned." Without qualification, we can tell you that the feature set that Orgasmatron 5.2 brings to the table for free runs circles around anything you could buy (then or now) in the commercial marketplace. And, at one time or another, we purchased every Nortel feature good money could buy. There’s one other difference. Orgasmatron 5.2 runs swimmingly on a $200 Atom-based PC that you can purchase at any Best Buy as well as hundreds of other stores including Amazon, NewEgg, and Buy.com. We paid more than $200 to provision an additional extension on our Nortel switch! You, of course, can add as many extensions as you like. De nada.

So, why a new version of Orgasmatron in only a few weeks? Well, it’s not security-related. In fact, there is nothing wrong with continuing on with Orgasmatron 5.1. Unfortunately, it relied exclusively upon SIPgate to make free Google Voice calls in the U.S. and Canada. And SIPgate required an invite using an SMS message from a U.S.-based cellphone. That pretty well knocked out all of our friends living outside the United States. Today’s version fixes that by letting anyone sign up for a free IPkall phone number in Washington state. All you need is a valid email address. The setup process is a bit more complex because IPkall doesn’t support registered connections to their servers. But we’ll walk you through the additional steps and, once completed, your server will be just as secure as the SIPgate approach we set up with Orgasmatron 5.1. And few, if any, Linux skills are required to set up or manage Orgasmatron 5.2. As we’ve noted previously, if you can handle slice and bake cookies, you’ve got the necessary skillset! Be aware this is about a one-hour project, and you need to track through the article carefully, or the entire house of cards comes down.

New Asterisk Security Model. Orgasmatron 5.2 maintains our design goal of running an absolutely secure Asterisk PBX from behind a hardware-based firewall with either NO INBOUND PORTS exposed to the Internet with SIPgate or an IP-address-restricted IAX port for IPkall. Don’t defeat this security mechanism by exposing additional ports on your PBX in a Flash server to Internet access. And choose your NAT-based firewall/router carefully. All of these devices are not created equally. Not only do some perform better than others, but certain models are notoriously bad at handling NAT-based routing tasks, a critical requirement in the Asterisk VoIP environment. In almost every case of problems with one-way audio, the real culprit can be traced back to a crappy router. For $35, you really can’t go wrong with the dLink WBR-2310. If you want traffic shaping functionality as well, take a look at dLink’s Gaming Router, our personal favorite.

As long as your router, Google Voice, SIPgate, and IPkall passwords are secure, you can sleep like a baby. We use an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada because Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you’re calling is connected. The SIPgate or IPkall trunk is used by Google Voice to call you back so the inbound call is always free. We handle the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just refrain from using extension 75 in Asterisk for personal conferencing!

The choice is yours. You can use SIPgate with no incoming ports exposed to your server from the Internet. Or you can use IPkall and map UDP port 4569 (IAX2) on your hardware-based firewall to the internal IP address of your new PBX in a Flash server. Even with the IPkall setup, we’ve locked down IPtables (our Linux firewall) to restrict IAX access to several specific IP addresses so your server remains absolutely secure. We’ve also included support for FonicaTec’s IAX offering for those that want a backup IAX provider. We’ll have much more to say about IPtables in coming weeks.

If you’ve already installed Orgasmatron 5.1 and it’s working for you, do you need to upgrade? NO. With the exception of the new IAX support for IPkall, the code in Orgasmatron 5.2 is identical.

We, of course, continue to recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.

Swiss Army Knife Inventory. There’s no need for a Swiss Army Knife if you don’t know what all the blades are for. So, for those that are wondering what’s included in the Orgasmatron 5.2 build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.

A2Billing (/root/nv/install-a2billing)
Amazon S3 Cloud Computing
AsteriDex
CallerID Superfecta (FreePBX Module)
CallWho for Asterisk
Cepstral TTS (/root/nv/install-cepstral.sh)
Preconfigured Email That Works with SendMail
Extensions (16 preconfigured)
Fax Module using nvFax
FONmail
FreePBX Backups
Gizmo5 (Free Calls to Gizmo5 users worldwide: 1747xxxxxxx*1089)
Google Voice (preconfigured)
Hamachi VPN (/root/nv/install-hamachi.x)
Hotel-Style Wakeup Calls (FreePBX Module)
ISN: FreeNum SIP Calling from Any Phone
MeetMe Conference Bridge (just dial C-O-N-F)
Mondo Full System Backups (/root/nv/install-diskbackup.x)
NewsClips from Yahoo
ODBC Database Support
PogoPlug Cloud Computing
Reminders by Phone and Web
SIP URI Outbound Calling (call any SIP URI worldwide for free)
TeleYapper
Tide Reports with xTide
Trunk Lister Script (/root/nv/trunks.sh)
Trunks (Vitelity, Fonica, SIPgate, IPkall, and ENUM)
Twitter Interface (Make Free Calls and Send SMS Messages)
Weather by Airport Code
Weather by ZIP Code
Worldwide Weather
Zaptel Updater (/root/nv/zaptel-update.sh)

Prerequisites. Here’s what you’ll need to get started:

Broadband Internet connection
Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
$200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
Free Google Voice account (HINT: Under $2 on eBay)
Free SIPgateOne residential account (Use cell to get SMS invite) OR
Free IPkall IAX account

Learn First. Install Second. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today’s VoIP world. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.

Today’s Drill. The installation process is straight-forward, but a little different than the Orgasmo 5.1 scenario because of the need to accommodate IPkall. Just don’t skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:

1. Install the latest version of PBX in a Flash
2. Run the Orgasmatron 5.2 Installer
3. Configure a softphone or SIP telephone
4. Configure Providers for Orgasmatron 5.2
5. Enter your Google Voice and SIPgate/IPkall credentials
6. Change existing passwords to secure your system

Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta on a new Atom-based PC. This beta is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use the identical payload for versions 1.3, 1.4, 1.5, and 1.6 of PBX in a Flash. The beta label simply means we haven’t had time to sufficiently test CentOS. But this is not a Microsoft-style beta so fear not!

Download the 32-bit, PIAF 1.6 version from SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot from the installation CD and type ksalt to begin.

WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can’t find the kickstart file. Just tab to OK and press Enter. Don’t change the name or location of the kickstart file! This will get you going. Think of it as a CentOS ‘feature’. 🙂

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes

When prompted, change the ARI password to something really obscure. You’re never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source on the Orgasmatron builds because of parking lot issues in the latest releases of Asterisk.

Running the Orgasmatron 5.2 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.2 installer:

cd /root
wget http://pbxinaflash.net/orgasmo52.x
chmod +x orgasmo52.x
./orgasmo52.x

Have another 15-minute cup of coffee. It’s a great time to consider a modest donation to the Nerd Vittles project. You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Now run passwd-master¹. Set your FreePBX passwords to something very secure but different from your Linux root password.

Next, type status² and press Enter. Write down the IP address of your new server.

If you’re using IPkall, now’s the time to log in to your hardware-based firewall/router and map UDP port 4569³ to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don’t worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you’re using SIPgate instead of IPkall.

For good measure, we recommend you reboot your server at this point. The command to type is simple: reboot⁴

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone, and you’ll find lots of recommendations on Nerd Vittles. For today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 82812661 as the password for extension 701 and the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Don’t Forget! After you change your extension passwords later in this tutorial, you will need to update the password entry in X-Lite, or you will no longer be able to place calls! In fact, you will get locked out of your server for 90 minutes after three failed password attempts. So put this on a sticky note so you don’t forget, or you’ll regret it in about 15 minutes.

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. And there’s really no reason you can’t use both if they’re available in your location. Do NOT use special characters in your provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we’d recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You’ll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don’t worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn’t matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you’ll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You’ll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you’ve opted to use IPkall, here’s the drill. First, you’ll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we’ll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don’t have an available number, your request apparently goes in the bit bucket. You’ll know because IPkall typically turns these requests around in a few minutes. Don’t worry about the mothership entry. We’ll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it’s dynamic, then you’ll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you’ve got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you’ll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It’s worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

We’re going to be making some entries in FreePBX which is the web-GUI that manages PBX in a Flash. For now, we simply need to enter your new IPkall phone number so that incoming calls to your IPkall number will actually ring on your softphone. Later, we’ll make some further adjustments once we get Google Voice humming along.

Using a web browser from your desktop, log in to FreePBX 2.6 at the following link substituting your server’s private IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master).

When FreePBX loads, choose Setup, Trunks, ipkall (iax). In the USER Context field, enter your 10-digit IPkall phone number. Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it’s a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Now let’s test your new phone number. Call your IPkall number from a cellphone or some other phone. Your softphone should ring. Answer the call, and be sure you have voice in both directions! Do not proceed without success here, or the rest of the adventure is a waste of your time.

Configuring Google Voice. Google Voice still is by invitation only so the first thing you’ll need is an invite. If you’re in a hurry, then stroll over to eBay where you’ll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you’ve chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.

Google then will place a call to your number and ask you to enter a confirmation code that’s been provided. When your cellphone (SIPgate) or softphone (IPkall) rings, answer it and punch in the number. Wait for confirmation. Then hang up.

As we mentioned earlier, there’s no reason you can’t set up both SIPgate and IPkall forwarding numbers in Google Voice. Just repeat the drill with the other provider’s number if you wish to activate both numbers for use with Google Voice. They’re not both going to ring simultaneously as you will see in a minute.

While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

Call Screening – OFF
Call Presentation – OFF
Caller ID (In) – Display Caller’s Number
Caller ID (Out) – Don’t Change Anything
Do Not Disturb – OFF

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Finally, place a test call to your new Google Voice number and be sure your cellphone or softphone rings. Don’t move forward until you’ve been able to successfully place a call to your phone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to PBX in a Flash. We’re ready to insert your Google Voice credentials and SIPgate/IPkall number into PBX in a Flash. You’ll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate or IPkall RingBack DID (beginning with a 1). Don’t get the 10-digit GV number mixed up with the 11-digit SIPgate/IPkall RingBack DID, or nothing will work. 🙂

Log back into your server as root and issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

Configuring FreePBX. Now shift back to your Desktop and, using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master). Depending upon which intermediate provider you’re using, do the following:

SIPgate Setup. When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:

7004484f0:B8TTW3@sipgate.com/4155201234

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

SIPgate and IPkall Setup. While still in FreePBX with your browser, click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate or IPkall Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

Securing FreePBX. You’re almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.

Don’t forget to adjust your X-Lite password to match the password entry you made for extension 701!

Orgasmatron Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let’s place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then, in the subsequent articles below, we’ll cover in detail how to customize every application that’s been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we’d recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

The VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. Using your new softphone, begin your adventure by dialing these extensions:

D-E-M-O – Nerd Vittles Orgasmatron Demo (running on your PBX)
1234*1061 – Nerd Vittles Demo via ISN FreeNum connection to NV
17476009082*1089 – Nerd Vittles Demo via ISN to Google/Gizmo5
Z-I-P – Enter a five digit zip code for any U.S. weather report
6-1-1 – Enter a 3-character airport code for any U.S. weather report
5-1-1 – Get the latest news and sports headlines from Yahoo News
T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
F-A-X – Send a fax to an email address of your choice
4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
M-A-I-L – Record a message and deliver it to any email address
C-O-N-F – Set up a MeetMe Conference on the fly
1-2-3 – Schedule regular/recurring reminder (PW: 12345678)
2-2-2 – ODBC/Timeclock Lookup Demo (Empl No: 12345)
2-2-3 – ODBC/AsteriDex Lookup Demo (Code: AME)
Dial *68 – Schedule a hotel-style wakeup call from any extension
1061*1061 – PBX in a Flash Support Conference Bridge
882*1061 – VoIP Users Conference every Friday at Noon (EST)

Click above. Enter your name and phone number. Press Connect to begin the call.

Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you’ll want to activate CallerID Lookups in FreePBX. Choose Setup, CID Superfecta, Default and enter the maint password you created with passwd-master. Then choose Tools, Module Administration, CallerID Lookup, Enable, Process and Save the Settings. Then edit each of the Inbound Routes and choose CallerID Superfecta as the CID Lookup Source. Save your changes. Finally, choose Setup, CallerID Lookup Sources, CallerID Superfecta and be sure your maint password created with passwd-master is correct here, too. If not, update it. For additional tips, visit the forums.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you’ve heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It’s also perfect for off-site backups!

Also check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches for Orgasmatron 5.1 and 5.2. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.

Coming Attractions. In our next episode, we’ll walk you through the process of adding a second, third, fourth, and fifth Google Voice line to your server so that you’ll never run out of free calling on your server. Enjoy!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.

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…

passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. [↩]
status is the PIAF utility program that displays the current status of most major applications running on your server. [↩]
Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device "inside" your firewall for further processing. [↩]
reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. [↩]

Is It Time to Get Your Buzz On?

Wednesday, February 17, 2010 / 1 Comment on Is It Time to Get Your Buzz On?

If you’ve been on vacation for the past week without your computer and cellphone, welcome back. Here’s what you missed: Everything! Yes, in seven little days, we’ve watched in amazement as Google the Giant Internet Gorilla with unlimited financial resources jumped off the 1,000 foot cliff without a parachute and then set about discovering how to land safely. Google jumped head first into social networking with a new product, Google Buzz, and decided to kick start the project using the age-old business trick of tying the new product to an old one, in this case their 175 million existing Gmail customers.

In the process of introducing Google Buzz, Google got almost everything wrong. Ironically, it’s the only new Google product that’s ever been introduced without the beta label. We’ll let you be the judge of whether Buzz is ready for prime time. Consider the following. There was the exposure of people’s most cherished Little Black Books to complete strangers. And cross-scripting security issues have been reported with the potential of exposing users’ Google credentials which in turn provides the key to the Google Checkout castle not to mention all of your most confidential emails. Then there were complaints that customer’s geolocation data was being compromised without user authorization. If that wasn’t enough on your plate for one week, Google now has to contend with a class action lawsuit and several government investigations into its Buzz business practices. Aside from that, did we mention Google Buzz is a huge hit!

If social networking is your thing, then you’re going to love Google Buzz. Think of it as FriendFeed on Steroids. Rich multimedia and location-based services on top of everything you always loved about IRC. And, if you have an Android 2.0+ phone, it gets even better with complete integration into Google Maps 2.0.

Unfortunately, everyone has been so busy with damage control and Google bashing that there’s been precious little time to actually explore the potential for Google Buzz in the social networking community. Our brief look at the product and its potential suggests that Google has another winner on its hands. It’s just too bad it wasn’t introduced in a manner similar to Google Wave so that users (and Google) could walk before attempting to fly. Here’s our first crack at how Google Buzz could actually be integrated into the blogging community, in our case making Google Buzz an integral part of a WordPress blog. And there’s also the widget approach from MoreTechTips.net which we’ve tweaked in the right margin below our Google maps. We’ll have more to say about these methodologies in coming weeks. In the meantime, come join the fun and Get Your Buzz On.

Nerd Uno’s Latest Buzz

Some Recent Nerd Vittles Articles of Interest…

Introducing PogoPlug: Cloud Computing for $100 per Terabyte

Tuesday, February 9, 2010 / 18 Comments on Introducing PogoPlug: Cloud Computing for $100 per Terabyte

Ever wished you could build and manage your own Cloud Computing Center with minimal cost and no recurring charges… ever? Well, today’s your lucky day.

It takes a lot to get us excited about a new product offering. But this one is a real winner! For under $130, Cloud Engines provides you your very own PogoPlug 2.0 device that connects to your router and shares up to four USB drives over the Internet. At today’s prices and ignoring sales tax, that means you can put eight terabytes of Cloud Storage on line for a one-time cost of about $100/terabyte. To give you a point of reference, Google will rent you the same space for $256/terabyte… per year. And Google is one of the least expensive Cloud Computing resources out there. Here’s the math for naysayers:

4 – WalMart¹ 2TB WD MyBook Drives @ $169 each = $676
1 – PogoPlug 2.0 Device @ $129 each = $129
ONE-TIME, NON-RECURRING COST: $805/8TB or $100/TB

For those that don’t need 8 terabytes, the 2 terabyte setup including the drive and PogoPlug device is still just over half the one-year rental rate of equivalent storage from Google. And, just to be clear, this isn’t merely a storage device (like Amazon S3) requiring downloads before the files can actually be used. PogoPlug’s software makes these USB drives an integral part of your Desktop just like any other attached storage devices. Think WebDAV! So it makes a perfect home for your music, movie, and photo collections. There also are loads of Open Source applications for PogoPlug for those that like to tinker. And you can use PogoPlug to keep synchronized backups of your important files.

Other Options. Be aware that for about $50 less, you can purchase the Seagate FreeAgent DockStar Network Adapter which includes a single year of PogoPlug Internet support. After that, it’s $30 annually. Translation: By the end of the second year, you’re better off with the PogoPlug. So the choice is a No-Brainer in our book. But, the fact that Seagate is also standing behind the PogoPlug design should make everyone sleep more soundly.

Deployment. After a one-minute, one-time setup over the Internet, you can securely access all of your USB drive resources via PogoPlug using either a web browser or one of several free desktop applications that are available for Windows, Mac OS X, Linux as well as Android phones, iPhones, and (earlier today) Blackberrys. And you get free support and a terrific forum. The device works flawlessly behind either a DSL or cable modem AND a NAT-based router so there are no firewall issues to address. Just enter the serial number on the bottom of your device when you access the PogoPlug web site, and configuration is automatic.

Uploading Files. One of PogoPlug’s slickest features is its automatic cataloging of files which are uploaded. Once uploaded, you can view your Music, Movies, and Pictures by simply clicking on one of the buttons. Photos are cataloged into directories by the month in which the photos were taken. Music is indexed by artist, album, and genre. In addition, music by artist, album and genre as well as photo albums can be shared by entering email addresses for those that can access the materials, by enabling public viewing (assuming you have legal rights to do so), or by sharing items using your Twitter, Facebook, and MySpace credentials. We’ve shared a photo album just to give you an idea of how this works. The security and logistical nuts and bolts all are managed by Cloud Engines’ servers. You can review and modify the materials you’re sharing by clicking on the Files I Share link in your browser. Finally you can automatically alert those with share privileges when folder content is updated. Very slick!

Give PogoPlug a try. By clicking on one of our links, you also help support the Nerd Vittles project. We think you’ll be as thrilled as we are with this terrific new creation. Enjoy!

Need help with Asterisk®? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.

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…

The in-store pricing at WalMart is actually cheaper than on line for these particular drives. [↩]

Apple’s iPad: A Home Run for Education

Monday, February 1, 2010 / 8 Comments on Apple’s iPad: A Home Run for Education

We’ve been anything but a cheerleader for Apple lately. And that applies in spades to the iPad. If you follow us on Twitter, here’s a sampling of our comments since the iPad was introduced last week.

The Computer Illiterate’s Dream Machine: Meet the iPad. Thinking you’ll slip your existing AT&T or T-Mobile SIM into Apple’s new iPad? Think again.

iPad: The Good http://tr.im/ipadg, The Bad http://tr.im/ipadb, and The Ugly http://tr.im/ipadu

iPad: Uh, but wait, No Camera http://tr.im/ipadc, No Flash http://tr.im/ipadf, No Multitasking http://tr.im/ipadm. No thanks.

Funny: Hitler’s take on the iPad. http://tr.im/ipadah

Why Apple Doesn’t Want Flash on the iPhone and iPad? It Finally Makes $ense. http://tr.im/theflash

The Vote That Really Matters: A 16-year-old’s view of Apple’s iPad: iFail http://tr.im/ipad16 (via @scobleizer)

Michael Dell demos what the iPad coulda/woulda/shoulda been. http://bit.ly/czYPww (via @engadget) #android Mini5

iPad Web Surfing: Here’s what the future holds. http://tr.im/noflash (via @gadgetweb) #surfsdown

RT @cultofmac: "Pundits On The iPad’s Closed System: It’s Doom For PCs, No It’s Great" http://bit.ly/cpFV4v

Apple iPad Micro SIM guarantees that you’ll be paying for two wireless data plans instead of one. http://bit.ly/bYipZP

Funny: iPad v. A Rock http://bit.ly/b50XP2 (via @TechCrunch)

RT @TechmemeFH: Apple reinventing file access, wireless sharing for iPad (Prince McLean/AppleInsider) http://bit.ly/awHJzG

Today’s Math Lesson: Hulu + Flash = Free Internet Movies. iPad – Flash = Megabucks for Apple from iTunes Movie Store sales

RT @dcagle: The library of the future, courtesy of the iPad http://bit.ly/bFvDAE #apple #toon

So what’s with the headline? Have we changed our mind? Well, no. It’s a lousy machine for us and for anyone above the age of puberty. But sometimes you need to look beyond the forest to find the nugget in the trees. And we’ve found the iPad’s Sweet Spot: It’s Lower School Education, Stupid!

For all the reasons that make the iPad an undesirable computing device for adults, it turns out these same qualities make it an almost perfect learning platform for young children, ages 3 to 12. In fact, we think it has the potential to revolutionize preschool and elementary education.

For openers, we can all probably agree that the key to a good education is good teachers. And that’s especially true when it comes to computer education. The problem, of course, is that teachers of young children don’t have the time or the resources to keep up with computer technology because they’re so busy doing all the things that parents should actually be doing to raise their kids. So, other than turning kids loose with a computer game, PCs have been all but worthless in lower school education because the teachers never had time to master the devices themselves. The iPad fixes that because of its incredibly simple learning curve. Any teacher can master the richness of the iPad interface in an hour. And it turns out that’s probably true for young children as well. If you don’t believe it, hand a kid your iPhone and come back in an hour.

A computer is important in early education because it’s much more patient and individually focused than any teacher ever could be. A computer doesn’t care how many times it takes a kid to master a specific topic. And, for young children, they need the repetition at their own pace until they actually get it. The iPad can handle all of these repetitive tasks while freeing the teacher up for observation and pinpoint coaching. So it levels the playing field by getting the "slow learners" up to speed without the usual frustrations of dealing with kids with different levels of comprehension. And the iPad accomplishes this while making education fun instead of frustrating!

Young kids learn with their hands. Walk in any lower school classroom if you don’t believe it. The iPad is a hands-on device. You use your hands literally for everything: a mouse, a navigation instrument, a drawing tool, and for writing and typing. So it’s a natural for kids, just like a hammer.

If you’ve ever visited a Montessori school, you’ll come away appreciating how critically important group collaboration can be to early education. Working in teams enhances learning in so many ways. The iPad is a natural collaboration tool. It can be used to encourage kids to jointly develop rich multimedia reports pulling from the web, their textbooks, images, and their classmates. iWork for iPad at $9.95 per application is the perfect development tool. And, as Steve Jobs demonstrated, the iPad makes a perfect presentation tool. Teaching kids to stand in front of their peers and tell a story is probably the single most important thing kids can learn in elementary education. You learn a lot more teaching others than you’ll ever learn as a student. Most of today’s adults never got it… nor did they have the opportunity that the iPad presents.

We could write a book about the advantages which would flow from getting rid of hardback books. Not only would it save trees and natural resources, but it also could turn books into living, breathing educational tools with rich multimedia presentations instead of static images. Instead of kids lugging around a backpack full of textbooks which will be obsolete in a year or two, they could carry an iPad with all of their learning tools, their schedules, their homework, and their presentations. Think about the possibilities, and you’ll come to appreciate why the iPad really could revolutionize education as we know it. We hope so. Go talk to the educators in your community and get them excited about this Golden Opportunity. You’re only young once!

For a well-balanced, thought-provoking review of the iPad, head over to emergent by design.

We’ll leave you with Neil Curtis’ 3-minute, adjective-laced version of Steve Jobs’ iPad Introduction. And, just in case you missed the Grammy Awards last night, there was an iPad Presentation there as well. Funny stuff!

Some Recent Nerd Vittles Articles of Interest…

Tweet2Dial: SMS Messaging with Google Voice and Twitter

Tuesday, January 19, 2010 / 5 Comments on Tweet2Dial: SMS Messaging with Google Voice and Twitter

We continue our quest for convergence today by adding the missing piece to our recent Tweet2Dial application. In addition to free calls to everyone in the U.S. and Canada as well as complete management of your Asterisk® server from Twitter, today's enhancement lets you send SMS messages to any SMS device or cellphone in the U.S. and Canada using simple Twitter messages. And, best of all, Tweet2Dial is free and runs on almost any Asterisk or Linux server as well as every Mac on the planet.

Twitter already provides some basic SMS integration that allows you to use SMS messages to send tweets. You also can opt to receive some Twitter messages via SMS whenever your friends post a new Tweet. But Twitter's SMS functionality is Twitter-centric meaning that both you and your friend must be Twitter users to take advantage of the SMS enhancements. Tweet2Dial adds the missing piece so that you can send SMS messages to anyone with an SMS-capable device in the U.S. and Canada whether or not they have a Twitter account. After all, that's what convergence is all about!

If you've already installed Tweet2Dial, we'll walk you through upgrading your existing setup in this article. If you haven't previously installed Tweet2Dial, then all you need to do is read the updated, original article which now includes coverage of the SMS functionality. Keep in mind that current Twitter API call limitations still limit you to one call or SMS message or Asterisk CLI command per minute. We'll remove this limitation once Twitter expands the hourly API call restriction.

Upgrading Tweet2Dial. For those that already have installed Tweet2Dial, here are the steps to add the SMS functionality. Just log into your server as root and issue the following commands. For Mac users, there is no root account. Just open a Terminal window while logged in with the user account used to set up Tweet2Dial initially and skip the cd /root command below:

cd /root
mv tweet2dial.php tweet2dial2.php
wget http://pbxinaflash.net/source/twitter/tweet2dial.tgz
tar zxvf tweet2dial.tgz
rm tweet2dial.tgz

Now open your old Tweet2Dial application (renamed to tweet2dial2.php) and write down your existing settings. Then edit tweet2dial.php and plug your old settings back in to restore access to your Google Voice account, your Asterisk server (if desired), and your Twitter friends. That's it! You're finished.

Sending SMS Messages with Twitter. To send new SMS messages, you'll use the same scenario outlined in the original article to place free phone calls. Just send a direct message to your secondary Twitter account. Only those that you have authorized as friends can send direct messages to this account so it's as secure as you want it to be. The Twitter Direct Message syntax for an SMS message looks like this where 6781234567 is the 10-digit cellphone number or Google Voice number of the SMS recipient:

SMS:6781234567:Here is a sample SMS message

Any replies to an SMS message which you send using Twitter will be forwarded to the email address that you used to set up your Google Voice account. Enjoy!

Special Thanks. Our tip of the hat again goes to the Pygooglevoice Development Team: JEIhrig, justquick, jacob.feisley, and nagle. Without their pioneering work, there would be no Tweet2Dial, no Orgasmatron V, and no Googlified Messaging for Asterisk. Terrific code! Thank you.

Happy Birthday to Us! Well, today's the Big Day. Today marks the Fifth Birthday for Nerd Vittles. Seems like only yesterday. Thanks for putting up with us all these years!

Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.

whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.

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...

Tweet2Dial: Free Google Voice Calling & SMS with Twitter

Monday, January 4, 2010 / 11 Comments on Tweet2Dial: Free Google Voice Calling & SMS with Twitter

To celebrate the New Year, it seemed only fitting to bring Google Voice calling out of the cloud and into our favorite social hangout. For our special New Year's project, we're pleased to introduce Tweet2Dial. It lets you use Twitter or your favorite Twitter client to make free outbound calls through Google Voice to anyone in the United States or Canada. Just send a Direct Message to your new Twitter account and, in less than a minute, your phone will ring connecting you to the person's phone number you specified in your Twitter message. In addition, you also can send SMS messages to anyone with an SMS-capable device in the U.S. and Canada. All of this magic is managed on your existing Asterisk® server or almost any Linux server or Mac. There's no Asterisk overhead to process the calls and SMS messages because Asterisk isn't required! But, to start 2010 off on the right foot, we've included a little bonus at the end of this article for all the Asterisk administrators in the house. If you happen to be using an Asterisk server, you now can manage it from Twitter with Tweet2Dial, too.

For those with cellphone plans that let you designate certain numbers for free, unlimited calling (such as Sprint, AT&T, Verizon, and T-Mobile), adding your Google Voice number to your preferred number list will mean that all of your Tweet2Dial-originated cellphone calls to anyone and everyone throughout the U.S. and Canada will now also be totally free with no impact on your bucket of call minutes.

Yes, we know Jajah is working on something similar for Twitter. But you have to be invited to participate in Jajah's beta (we didn't make the cut!), free calls are limited to two minutes, and both parties have to have a Twitter account which doesn't work too well for calling grandma. So why put up with all the limitations and restrictions of Jajah when you can do it yourself?

There's been some tech chatter that the procedure we've outlined below is complicated. If you can paint by number or bake cookies from the back of a Nestle's bag, trust me. You can handle this! Getting a Mac or a Linux server set up to support Tweet2Dial only takes a minute or two. So ignore the trade rags. Some of them can barely read. 🙂

If you've already gone through our Google Voice tutorial which enables free Google Voice calling on your Asterisk server, or if you've installed our all-in-one Orgasmatron V build on your Asterisk server, or if you have a Mac or you've built your own Linux server without Asterisk, there's no need to wait for Jajah and no need to limit your calls to two minutes or to those with Twitter accounts! You can call anyone in the United States or Canada right now, talk as long as you like, and do it all for free with Tweet2Dial, Twitter, and Google Voice! If you're a Windows user, check out the Google Voice Dialer for Windows.

Prerequisites. To get started, you can use your Asterisk server configured for Google Voice as we've outlined above. We won't actually be using Asterisk to place the calls, but our previous tutorials get your server properly set up with Google Voice and the latest, awesomest¹ pygooglevoice to support Tweet2Dial. Any of the Asterisk aggregations such as PBX in a Flash will work great.

If you don't have a PBX in a Flash server with Google Voice already configured, shame on you! Just kidding. Actually, any recent CentOS or Fedora Linux server will work just as well today. Log into your server as root. Run rpm -q python to make sure you have at least Python 2.4 installed on your system. If not, run: yum update python. Then execute the following commands:

cd /root
yum install python-setuptools
easy_install simplejson
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice*
cd pygooglevoice-0.5
python setup.py install

Tweet2Dial also will run just fine on any Mac of recent vintage. We've actually tested it with Snow Leopard. Basically, to get Python and Apache set up properly, you have to enable root access, switch to root user access with su in Terminal, activate PHP support in Apache, turn on Web Sharing in System Preferences->Sharing, run easy_install simplejson as root to install simplejson (the Python Setup Tools already are in place!), using a browser download pygooglevoice to your Downloads folder, untar it as root in Terminal with the same command as above, and then while still logged into Terminal as root, go to the Downloads/pygooglevoice-0.5 folder and run the following command: python setup.py install. The only variations in the Tweet2Dial setup will be the storage location for Tweet2Dial (there is no root folder on a Mac) and the methodology for setting up the crontab entry (HINT: we'll run crontab -e to add a crontab entry since there is no /etc/crontab file). Just follow along using the Mac-specific instructions below for details, and everything will work swimmingly.

To test whether your server is properly configured for Tweet2Dial, log in as root and type: gvoice. You should be prompted for an email address. If so, press Ctrl-C to exit. You're ready to roll. If not, pygooglevoice has not been properly installed on your server.

You'll obviously need a Google Voice account. Request an invite here or just post a brilliant comment below, and one might magically appear in your inbox. Configure your Google Voice account with all the phone numbers from which you want to place outbound calls. One of these numbers will already be the go-between number for Google Voice and your PBX in a Flash server (IPkall or SIPgate) if you've followed our previous tutorials. Now simply add additional numbers that you want to use to place outbound Google Voice calls. This would include numbers such as your cellphone, your vacation home, and your direct-dial office number. You do not need to enable them for ringing when inbound calls arrive on your GV number.

For today's project, you'll also need a new Twitter account even if you already have one. Why? Because you can't send a Direct Message to yourself with Twitter. So we'll use your primary Twitter account to send Direct Messages with dialing instructions to your secondary Twitter account. Then we'll use Tweet2Dial to poll your secondary account and retrieve the dialing instructions to actually place the outbound calls with pygooglevoice through your server. It sounds harder than it actually is. Honest! Assuming you already have Google Voice running on your Asterisk server, you'll be tweeting away in 10 minutes. If you have a current Linux server, add an extra 2 minutes to install pygooglevoice using the steps above.

Usage Considerations. Before someone asks, let's address Question #1. Can others send messages to my Twitter account in order to make outbound calls through my server using Google Voice? And the answer is yes and no. We're going to configure your new secondary Twitter account with Protect My Tweets enabled. This means you have to approve friends and also become their friend before they could send a Direct Message to your secondary Twitter account. So, yes, if you approve, any Twitter user could theoretically place calls using your Twitter secondary account. For the average reader, we wouldn't recommend it for a couple of reasons. Here's why.

Google Voice only lets you link a handful of phone numbers to your GV account. So, for your friends to be able to place calls using your GV credentials, you'd have to forfeit one of your allotted quota of numbers for each person... or their phone would never ring to place the outbound calls. Yours unfortunately would! Remember, Google Voice always places two calls to complete a connection: one to you (using one of the phone numbers defined in your GV account) and one to the person with whom you wish to speak.

The other reason for not opening this up to other callers is that Google Voice limits your account to one outbound call at a time. If others are using Twitter to make calls using your GV credentials, it means you can't. And there's no mechanism for easily identifying when a call already is in progress. So our recommendation is to keep your secondary Twitter account private and set up Following and Follower linkage only with your primary Twitter account. This will mean that Direct Messages to your secondary Twitter account can only originate from your primary Twitter account. You can still place outbound calls to anybody, but others can't!

Having said all of that, we've designed Tweet2Dial so that you can allow others to use your secondary Twitter account to place Google Voice calls using their own GV credentials. This saves them the aggravation of setting all of this up, but it means they have to trust you enough to share their Google Voice credentials. After all, what are friends for? 😉 At the end of this article, we'll walk you through how to do this if you really have the urge. We would hasten to add that the actual processing load on your server is virtually zero so don't be deterred by performance concerns. Pygooglevoice sends the calling instructions to Google Voice, and then your server is completely out of the call loop. We've still limited outbound call setup to one call per minute, but these calls do not have any impact on Asterisk resources and only very minimal impact on your server. The only drawback to hosting Tweet2Dial for your friends is that, if five simultaneous Twitter messages are sitting in the queue, it would mean the last call request won't be processed until about 5 minutes after the Twitter message was sent. But, unless you have a bunch of extremely chatty friends, call request congestion shouldn't be a problem.

One final word of caution. Twitter currently permits a maximum of 150 Twitter API calls per hour per account. There is some good news. Within the next few weeks, this limit will be increased to 1500 per hour, but it hasn't happened yet. This application is designed to poll your secondary Twitter account once a minute to retrieve and then discard your oldest, existing Direct Message. So it uses 120 of your allotted 150 API calls per hour to work its magic. You are well advised NOT to run any third-party Twitter applications with this secondary Twitter account, or you will quickly exceed the current connection limitation. When the API limit is reached, it means none of your pending call requests would be processed until the next hour rolls around... at least until Twitter raises this connection limit. Once Twitter raises the API limit, we may revisit our code and eliminate the current one call per minute limitation. So stay tuned!

Creating A Secondary Twitter Account. First, let's get your secondary Twitter account set up. Go to twitter.com and create a new account with a very secure password! You must enter a different email address than the one used for your primary account. Use one you can actually access! Log into your new account and choose Settings. Scroll down to Protect my tweets and check the box by clicking on it. Save your settings. NOTE: This check box is critically important. It keeps the entire world from being able to access your server! There are other layers in the security model, but this one is VERY IMPORTANT so verify it twice! Now log back into your primary account. Then goto http://twitter.com/SecondaryAccountName and request access. You'll get a message that your request for access has been sent. Log out and back into your secondary account once again. Authorize your primary account name as a Follower. Now log out and back into your Primary Account. We'll use it to send a Direct Message to your secondary account in a few minutes.

Installation and Configuration. To install Tweet2Dial, log into your server as root and issue the following commands:

cd /root
wget http://pbxinaflash.net/source/twitter/tweet2dial.tgz
tar zxvf tweet2dial.tgz
rm tweet2dial.tgz

If you're doing this on a Mac, there is no wget application and no root folder so you'll need to download tweet2dial.tgz with your browser. Save it to your Downloads folder. Then open a Terminal window and execute this command:

tar zxvf Downloads/tweet2dial.tgz

Now let's configure the application:

nano -w tweet2dial.php

At the top of the file, you'll see the following lines:

// Your SECONDARY Twitter account username and password
$username = "TwitterUsername";
$password = "TwitterPassword";

// Authorized Twitter users with corresponding GV credentials go below
$user['twitname'][1]="YourPrimaryTwitterUsername";
$user['gvemail'][1]="YourGoogleVoiceEmailAddress@gmail.com";
$user['gvpass'][1]="YourGoogleVoicePassword";
$user['gvcall'][1]="6781234567";

// *** Leave everything below this line alone. 🙂

Begin by entering your secondary Twitter name and password by replacing TwitterUsername and TwitterPassword with your actual credentials. Be careful here. Capitalization matters! If you set up your Twitter username as gvNerdUno, don't enter gvnerduno! Now move down to the four $user entries. The first is your primary Twitter account name. Replace YourPrimaryTwitterUsername with your actual Twitter account name. Again be careful of capitalization! Next, enter the login email address for your Google Voice account replacing YourGoogleVoiceEmailAddress@gmail.com. Next, enter your Google Voice password replacing YourGoogleVoicePassword. Finally, enter one of the 10-digit ringback numbers you've configured in your Google Voice account by replacing 6781234567. Do NOT use the one that's reserved for use by Asterisk! This is the number that will be called by default whenever you place an outbound call with Twitter. You'll have the option of overriding it, but this saves your having to enter both a destination phone number and a callback number each time you wish to place a call. Be sure to preserve the quotes around each of the entries. Once you've double-checked all of your entries for typos, save your changes: Ctrl-X, Y, then Enter.

Tweet2Dial Test Drive. Now that everything is set up, let's place a test call to be sure everything is working. Log into your primary Twitter account. Click on Direct Messages. Choose your secondary Twitter account from the pulldown menu. In the block below Send a Direct Message, enter a 10-digit number in the U.S. or Canada that's different from your default callback number. Then click the Send button. It's that simple! Once Twitter tells you the message has been sent, log into your Asterisk server and execute the following commands.
cd /root ./tweet2dial.php
If you're on a Mac, just open a Terminal window and type ./tweet2dial.php. In either case, you should get a response indicating that your call has been placed, and your default phone number should begin to ring. When you answer it, Google Voice will place a call to the 10-digit number that you entered in your Twitter direct message above.

Now, just for fun, run Tweet2Dial again: ./tweet2dial.php. If everything is working properly, you will see the following message: Nothing to do.

Finally, assuming you have configured another callback number in Google Voice that is close at hand and not your Asterisk callback number, send another Twitter direct message with the following syntax: 8439876543:6781234567 where 8439876543 is the 10-digit number of someone you wish to call and 6781234567 is a 10-digit ringback number already set up in your Google Voice account. Once the message has been sent, run Tweet2Dial again from the command prompt.

When you're sure everything is working reliably, add the following entry to the bottom of /etc/crontab unless you're using a Mac. This will run the application once a minute around the clock looking for incoming Twitter messages:

* * * * * root /root/tweet2dial.php > /dev/null

If you're running this on a Mac, add an entry to your crontab like this. From the Terminal window, run: crontab -e. Once the vi editor opens, type:

* * * * * /users/youracct/tweet2dial.php

Substitute the name of your Mac account for youracct. Then press the Esc key followed by :wq. Check your work by typing: crontab -l. Your entry should look like this:

* * * * * /users/youracct/tweet2dial.php

Sending SMS Messages with Twitter. To send SMS messages using Twitter, you'll use the same scenario outlined above to place free phone calls. Just send a direct message to your secondary Twitter account. Only those that you have authorized as friends can send direct messages to this account so it's as secure as you want it to be. The syntax for an SMS message looks like this where 6781234567 is the cellphone or Google Voice number of the SMS recipient:

SMS:6781234567:Here is a sample SMS message

Any replies to an SMS message which you send using Twitter will be forwarded to the email address that you used to set up your Google Voice account.

For Whiz Kids Only. Now let's say you want to let your spouse use her Twitter account to place calls using her very own Google Voice credentials. First, you need to authorize her as a follower in your secondary Twitter Account. Second, you need to add a new block of code in tweet2dial.php that looks like the following. Place it immediately below the existing $user entries in the file:

$user['twitname'][2]="SpousePrimaryTwitterUsername";
$user['gvemail'][2]="SpouseGoogleVoiceEmailAddress@gmail.com";
$user['gvpass'][2]="SpouseGoogleVoicePassword";
$user['gvcall'][2]="6781234567";

// *** Leave everything below this line alone. 🙂

Notice that the only change is this array subset is numbered [2] while the original was numbered [1]. You can add as many as you like so long as you increment this number and provide the credentials for each user. Now you have your own little Jajah-like sandbox, and it's absolutely free.

For Asterisk Administrators Only. Want to manage your Asterisk server from Twitter? There's an app for that. We promised you a New Year's bonus so here it is. First, read our last article which explains how to manage your Asterisk server using email messages and the Asterisk CLI. Now you can do exactly the same thing using Twitter direct messages. The only Twitter user that can do this on your server is the Twitter account name you specified in the #1 $user slot above. So you don't have to worry about your pals trashing your Asterisk server if you give them privileges with Tweet2Dial. The syntax for issuing CLI commands using Tweet2Dial looks like this:

CLI: database show cidname 8437978000

Just be sure Direct Messages from your primary Twitter account begin with CLI in all CAPS followed by a colon, a space, and then the desired CLI command. That's all there is to it. You'll get a confirmation Direct Message in your main Twitter account once the command has been executed assuming you have established Following and Follower linkage between your primary and secondary Twitter accounts. Test sending DMs in both directions to double-check it. And if you've enabled email delivery for Direct Messages in your Twitter configuration, you'll get an email confirmation as well. Because of Twitter's 140 character limitation, some commands such as help don't provide all of the output you normally would receive from the CLI. You'll only get the last line. Aside from that, the CLI functionality is identical to interacting directly with the Asterisk CLI and the email implementation we outlined previously. Here's the CLI response:

Before you can use the CLI interface in Tweet2Dial, you have to enable it. Edit tweet2dial.php and change $CLIenable=false to $CLIenable=true. And, yes, we understand there are some of you that don't trust Twitter to keep your commands secure. Well, first of all, in order to penetrate your Asterisk server, someone would have to send a Twitter Direct Message from your primary Twitter account. So they'd need your password and they'd need to know the syntax for Asterisk CLI commands AND the syntax for sending them via Twitter. But, there's always a Cracker Rapper² somewhere. Right? So we've also built a password into the system at your server's end so you can sleep more comfortably. The default password is CLI. But feel free to change it to anything you like. Just edit tweet2dial.php and find this line: $CLIpword = "CLI";. Replace CLI (between the quotes only!) with whatever password you'd like. After saving your changes, you'll need to adjust your Twitter messages accordingly. For example, if you changed your password to FooBar, then your future Twitter CLI command syntax would look like this: FooBar: help. Enjoy!

Special Thanks. As Nerd Vittles prepares to celebrate its Fifth Birthday, we want to take a moment to thank those that have made Nerd Vittles and the PBX in a Flash project possible. Without the generous financial support of Vitelity and Google's AdSense program plus the unwavering support of our hosting providers who provide free downloads of PBX in a Flash around the globe, all of what we do would be much more difficult and expensive! It's not too late for you to kick in a nickel or two as well if a fleeting moment of generosity should strike. 😉 There's a Donate button at the top of the page. Finally, we want to thank Digium® for their continuing support of the Asterisk project and their generous contribution of hardware to the PBX in a Flash development team during 2009. Happy New Year everybody!

Need help with Asterisk? Visit the PBX in a Flash Forum.
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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.

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