Home » Posts tagged 'asterisk' (Page 60)

Tag Archives: asterisk

The Most Versatile VoIP Provider: FREE PORTING

The Incredible PBX: Adding Multiple Google Voice Trunks

About the only drawback to Google Voice's free U.S. and Canada calling with the Incredible PBX has been the fact that you could only make one outbound call at a time... at least on Google's nickel. So today we'll fix that, and you can enjoy simultaneous outbound calls using as many Google Voice trunks as you have signed up for. If you're in the U.S., you're eligible and no invitation is required. Just head over to the Google Voice site to register.

Today's Incredible PBX enhancement also will permit you to set up multiple inbound DIDs for different area codes across the country which may save your out-of-town friends and relatives a little change when they want to contact you. And to think we had $200 a month phone bills in our college days just to call the hometown honey. The wonders of modern technology!

Prerequisites. Here's what you'll need to get started today. First, you need a functioning Incredible PBX. So start by installing Incredible PBX. Second, you'll need a second Google Voice account. And finally, you'll need an additional SIPgate One number.

Installation Assumptions. We'll walk you through the steps to get a second account activated with the Incredible PBX. If you need more than two, just repeat the steps below and substitute a new number for 2 in every step. As with baking cookies, if you skip a step, the cookies taste like crap. 🙂 For security reasons, we're using an additional SIPgate One account for the second setup. This avoids having to open up SIP access in your firewall which would require additional locking down of IPtables to specific SIP IP addresses.

Setting Up New SIPgate and Google Voice Accounts. As was true with the initial Incredible PBX setup, the first steps in activating a second line are to create and configure your SIPgate account and then tie that number into your new Google Voice account. For ease of reference, we've repeated below the pertinent portions of the original Nerd Vittles article.

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 and working properly. 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 complete the configuration of The Incredible PBX. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring Google Voice. Once you've signed up for a new Google Voice account, choose a telephone number and plug in your new SIPgate number as the destination for your Google Voice calls and choose Office as the Phone Type.

Google Voice will place a test call to your number which SIPgate will forward to your cellphone. Enter the two-digit code that's displayed when you're prompted to do so.

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.

Once you've confirmed your Google Voice number, revisit SIPgate and remove all parallel calling numbers including your cell number. Be sure you've written down your SIPid and SIPpassword while you're there!

FreePBX Overview. Don't be intimidated by the FreePBX setup instructions which follow. All we're really doing is cloning the original pieces of information that made Google Voice work in the initial Incredible PBX setup. For most of the items, we'll just tack a 2 onto the names previously used. Nothing prevents your adding 3, 4, and 5 accounts down the road if you have additional Google Voice and SIPgate accounts to support each iteration.

To begin, use a web browser to open FreePBX on your Incredible PBX. Using the actual private IP address of your server, go to the following link: http://192.168.0.33/admin.

Adding Parking Lot Slots. As originally configured, the Incredible PBX provides 5 parking lot slots for use on your PBX. These are numbers that let you temporarily "park" calls so that they can be picked up on another extension. One of those slots (75) is used by the Incredible PBX to place outbound Google Voice calls. If you want the ability to place simultaneous outbound Google Voice calls using multiple trunks, then we need additional parking lot slots for each simultaneous call. We recommend bumping up the number of parking lot slots from 5 to 9. Then you can use 75-79 for up to 5 simultaneous outbound calls with Google Voice. Here's how. In FreePBX, choose Setup, Parking Lot, Number of Slots: 9. Your entries should look like this screen shot:

When you've made the change, click Submit Changes, Apply Configuration Changes, Continue with Reload.

Creating Additional Custom Destinations. You'll recall that Google Voice actually places two calls when you make an outbound call. First, Google Voice calls you back. Then Google Voice places a call to your desired destination. The callback to you is handled transparently in Incredible PBX using pygooglevoice and Asterisk®'s parking lot feature. To handle multiple simultaneous calls, you'll need additional custom destinations. Here's how. In FreePBX, choose Tools, Custom Destinations, Add Custom Destination. Then make your new entries for custom-park2 look like this:

When you've made the entries and carefully checked them, click Submit Changes, Apply Configuration Changes, Continue with Reload.

Creating Additional Inbound Routes. Now we need an additional Inbound Route to handle the second incoming call generated by Google Voice. Here's how. In FreePBX, choose Setup, Inbound Routes, Add Incoming Route, gv-ringback2. Make the entries shown in the screenshot below substituting your 10-digit SIPgate/IPkall and Google Voice numbers in the appropriate fields. Be sure to choose Custom GV-Park2 as the Custom Destination for this Inbound Route. Check your entries carefully, a typo here will kill completion of the calls!

When you've made the entries and carefully checked them, click Submit, Apply Configuration Changes, Continue with Reload.

Creating Additional Custom Trunks. With every telephony provider, Asterisk needs a Trunk. In the case of Google Voice, we need a Custom Trunk for each Google Voice number to be used on your Incredible PBX. Think of a trunk as the bucket where Asterisk dumps an outbound call for processing. Two calls require two buckets. Three calls, three buckets. And so on. Well, that's almost true. Some providers can handle multiple calls, but Google Voice doesn't. So we need to make two changes in your trunk setup. First, we'll adjust the original Custom Trunk for Google Voice and limit it to one simultaneous call at a time. Then, we'll add a new Custom Trunk to support the second Google Voice account. Here's how.

In FreePBX, choose Setup, Trunks. In the right column, you'll see a list of all your existing trunks. Click on the second entry that looks like this: local/$OUTNUM$@ (custom). Be sure the Custom Dial String looks like what is shown below. If not, choose another trunk until you find the right one. Then make an entry of 1 in the Maximum Channels field:

When you've made the entry and carefully checked it, click Submit Changes, Apply Configuration Changes, Continue with Reload.

Now we're ready to Add the additional Custom Trunk. In FreePBX, choose Setup, Trunks, Add Custom Trunk. Make your entries look like what's shown below:

When you've made the Maximum Channels and Custom Dial String entries shown above and carefully checked them, click Submit Changes, Apply Configuration Changes, Continue with Reload.

Creating Additional Outbound Routes. FreePBX uses Outbound Routes to do just what the name implies: to route outbound calls to their destination. Outbound Routes are processed in the order in which they appear in the FreePBX Outbound Routes listing. We need to make three changes in the Outbound Routes processing to support a second Google Voice call path. First, we want to modify the existing Default Outbound Route to accommodate the second Google Voice account. Second, we want to add a new Outbound Route for the second Google Voice account so that calls can be placed directly with this route using a different dialing prefix. You'll recall that Google Voice calls in the Incredible PBX can optionally be dialed using the 48 prefix followed by a 10-digit number. The 48 spells GV on the phone key pad. So we'll add a new Outbound Route with a 482 (GV2) prefix which will tell Asterisk to route these calls out using the second Google Voice account. These prefixes can be anything you desire incidentally. Third, we'll need to move this new route UP the routes list so that it appears above and gets processed before the Default route. Here's how.

In FreePBX, choose Setup, Outbound Routes, Default. In the blank Trunk Sequence pulldown, choose the following entry: local/$OUTNUM#@custom-gv2. Now click the Add button. This should leave you with 3 outbound routes numbered 0, 1, and 2. Be sure your entries match the following:

When you've made the entry and carefully checked it, click Submit Changes, Apply Configuration Changes, Continue with Reload.

Now we're ready to add a new Outbound Route to support a custom dialing prefix for the second Google Voice account. In FreePBX, choose Setup, Outbound Routes. In the Add Route form, make the following entries:

When you've made the entries, click Submit Changes, Apply Configuration Changes, Continue with Reload.

Finally, look at the listing of Routes in the Right Margin. Using the arrow beside GoogleVoice2, move it up until it is just beneath the GoogleVoice entry. Then click Apply Config Changes, Continue with Reload.

Adding Additional SIPgate Trunks. If you set up your Incredible PBX originally using IPkall, then there already will be a sipgate trunk that can be used for this second line. Otherwise, you'll need to create a new sipgate2 trunk and clone the setup from the original sipgate trunk. Within FreePBX, goto Setup, Trunks and either Add a new SIP trunk or edit the existing sipgate trunk if it isn't already in use. If this is a newly added trunk, enter sipgate2 as the Trunk Name. The PEER Details under Outgoing Settings should be added so they look like this (substituting your actual SIPid and SIPpassword that were obtained from the SIPgate registration page:

type=peer
username=SIPid
fromuser=SIPid
secret=SIPpassword
context=from-trunk
host=sipgate.com
fromdomain=sipgate.com
insecure=very
caninvite=no
canreinvite=no
nat=yes
disallow=all
allow=ulaw&alaw

Blank out any data that's entered in the Incoming Settings section of the form. Then enter a Registration String with your actual SIPid, SIPpassword, and 10-digit SIPgate phone number:

SIPid:SIPpassword@sipgate.com/SIPphonenumber

Check your entries carefully for typos. Then click Submit Changes, Apply Configuration Changes, Continue with Reload.

Now is a good time to check and be sure the new SIPgate trunk registered with SIPgate. In FreePBX, choose Tools, Asterisk Info, SIP Info. Your newly created SIPgate trunk should display as Registered. If it says Request Sent, then you've got a typo in your credentials.

That takes care of all the FreePBX settings needed to support a second Google Voice number. Now we just need to add a chunk of dialplan code to Asterisk and restart Asterisk. Then you'll be ready to go. All of this is handled by a simple Nerd Vittles script so... not to worry! It's easy.

Adding Dialplan Code for Additional Trunks. Log into your server as root, and issue the following commands to download and run the dialplan configuration script. For future reference, be advised that there are configuration scripts for gv2, gv3, gv4, and gv5 with corresponding names.

cd /root
wget http://incrediblepbx.com/configure-gv2
chmod +x configure-gv2
./configure-gv2

When prompted, enter your 10-digit Google Voice phone number, your Google Voice email address, your Google Voice password, and your 10-digit SIPgate RingBack number. Check your work and then press the Enter key to adjust your dialplan and reload Asterisk. You now have a 2-line Incredible PBX. Enjoy!

The Incredible PBX: Basic Installation Guide

Adding Skype to The Incredible PBX

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

Adding Remotes, Preserving Security with 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...

The Incredible PBX: Adding a Free Skype Gateway to Asterisk

Last week we got The Incredible PBX all set up with free worldwide SIP calls, free U.S./Canada PSTN calls using Google Voice with SIPgate or IPkall, and rock-solid Asterisk® security using our new Zero Internet Footprintâ„¢ design. Because of licensing restrictions, we couldn't include Skype out of the box. If you're an individual and not a business, today we'll walk you through adding free Skype calling worldwide to your Incredible PBX. With today's addition, the Incredible PBX now provides free calling to nearly a billion phones around the world via Skype, SIP, ENUM, FreeNUM, and U.S./Canada PSTN connections. Yowza!

If you use the recommended hardware, today's setup procedure takes less than 10 minutes! Once it's complete, inbound and outbound Skype calling is totally transparent on your Incredible PBX. To reach a Skype number, just dial * plus the user's Skype name from any phone with an alphanumeric keypad. To place a Skype Out call (fees apply), dial 8 plus the user's area code and number. When your 500 million friends on Skype contact you using your Skype name, all of your Incredible PBX phones will ring just like any other inbound call. What's the difference in today's solution and Digium®'s commercial Skype for Asterisk product? For openers, our solution is $66 cheaper. It's free! And, if you're an individual, you won't need Skype's commercial Business Control Panel to make calls. Functionally, the results with your Incredible PBX Skype implementation are identical.1

To make the Skype Magic work, you'll need three pieces of software in addition to The Incredible PBX obviously: Sun's 6u12 Java SE Development Kit, Skype's Static Edition for Linux plus an existing Skype account, and Greg Dorfuss' SipToSis product which manages the Skype Gateway to Asterisk.

As far as hardware is concerned, we're assuming you're using our recommended $200 Acer Aspire Revo to host your Incredible PBX. With other hardware, your mileage may vary because CentOS 5.4 may or may not support your audio card and graphics mode with your video card. Both are required to get Skype working properly under X-Windows. If you have problems with some other type of hardware, take a look at the tips in our previous article on Setting Up a Skype Gateway to Asterisk as well as the comments. Better yet, visit your neighborhood Best Buy and purchase an Aspire Revo for a hassle-free install.


Installing JDK. Using your favorite browser, go to Sun's 6u12 Java SE Development Kit website, choose Linux for the platform, and agree to the license. Click Continue. Download jdk-6u12-linux-i586-rpm.bin and copy it to the /root directory of your Incredible PBX. Next, make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin). Then run it: ./jdk-6u12-linux-i586-rpm.bin. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Check to be sure Java was properly installed with this command: rpm -q jdk.

Installing Skype and SipToSis. Now we're ready to load the remaining components. While still logged into your Incredible PBX as root, download and run the skype-setup script2:

cd /root
wget http://incrediblepbx.com/skype-setup
chmod +x skype-setup
./skype-setup

Activating Your Skype Gateway. Now we're ready to place your Skype gateway in production. You'll need to perform these steps from the console on your Incredible PBX since we have to run Skype in graphics mode. This may look complicated. It's really not. It's just a bit tedious to figure out the sequence of steps, but we've done that part for you.

WARNING: Be sure that you use a dedicated Skype account on this server! Do not run the same Skype account on any other server or desktop, or it fails!

1. Start up X-Windows: xinit3

2. Start up Skype. While still logged into your server as root, issue the following commands:

cd /root/skype/skype_static-2.0.0.72
./skype

Now log in to Skype with your Skype name and password. Be sure to set Skype to autologin whenever it is started. Then, in the Skype configuration option, set Skype to always run minimized. Save your settings.

Place a Skype Test Call4 to echo123 to be sure your audio settings are set correctly. Again, with the Aspire Revo, this won't be a problem assuming you have plugged in a microphone and speakers. These can be disconnected after you're sure things are working properly. HINT: Intel Atom-based motherboards are a piece o' cake!

Once you've got Skype working and all of the Skype settings configured above, shut down Skype.

3. Restart Skype in Background Mode: ./skype &

Be sure to write down the PID for Skype in case you need to kill the job if something goes wrong. 🙂 If you forget the PID, you can obtain it with this command: pgrep skype. You can kill Skype with the following command using your actual PID instead of 12345: kill 12345.

4. Start up SipToSis: Press Enter if the command prompt doesn't reappear. Then...

cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype: yes. Important: Be sure to select the Checkbox to save this setting for future connections!

5. Testing Skype. Go to a softphone (X-Lite recommended!) connected to an extension on your Incredible PBX and dial *echo123. You should be connected to the Skype Call Testing Service. Try *nerdvittles for the Nerd Vittles Demo.

Assuming you have a little money in your Skype Out account, go to any extension connected to your Asterisk server and dial 8 + your home phone number. This will place the outbound call through SkypeOut at 2¢ a minute.

Reboot your server when you're sure everything is working properly.

GUI Tips. Here are a few navigation tips for managing your Asterisk console on your Incredible PBX:

1. Ctrl-Alt-F2 gets you a new login prompt for your server

2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To decipher your SipToSis PID: pgrep -f SipToSis. To kill SipToSis: kill pid# (that you wrote down). To kill Skype: kill pid# (that you wrote down). To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_linux

3. Ctrl-Alt-F9
gets you to the Asterisk CLI.

Automating the Skype Gateway Startup. Once everything is working reliably, reboot your server again, log in as root, and issue the command: /root/skype-start. Place a test call again using a softphone on your Incredible PBX. If everything works fine, you now can add the skype-start command to your server's startup script, and you're all set.

echo "/root/skype-start" >> /etc/rc.d/rc.local

Setting Up Speed Dials for Skype Friends. One of the wrinkles with Skype is that Skype uses names for its users rather than numbers. If you don't have a SIP URI-capable softphone, there's still an easy way to place calls to your Skype friends using FreePBX. Just add a Speed Dial number to your FreePBX dialplan. Choose Extension, then select the Custom type, provide an Extension Number which is the Speed Dial number (this could actually spell your friend's name using a TouchTone phone), enter a Display Name for your friend, and add an optional SIP Alias. Then insert the following in the dial field replacing joeschmo with your friend's actual Skype name. Save your entries and reload the dialplan when prompted.

SIP/joeschmo@127.0.0.1:5070

Security Warning. Do NOT expose UDP port 5070 to the Internet by opening a port on your hardware firewall. You do not need UDP 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your server!

Enjoy!

Update: As of May 1, you now can set your Google Voice number as your Skype CallerID number. Previously, Google Voice blocked the verification SMS messages, but no longer. Thanks, @zsafwan.

Adding Multiple Google Voice Trunks to The Incredible PBX



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...

  1. Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. []
  2. Here are the actual commands in the skype-setup script if you'd prefer to execute them one at a time:

    cd /root
    mkdir skype
    cd skype
    wget http://www.skype.com/go/getskype-linux-beta-static
    tar jxvf skype_static*
    yum install xorg-x11-server-Xvfb
    yum install qt4
    yum install xterm
    yum install libXScrnSaver.i386
    wget http://pbxinaflash.net/source/skype/siptosis.tgz
    cd /root
    wget http://incrediblepbx.com/skype-start
    chmod +x skype-start
    cp skype-start skype/.
    cd /
    tar zxvf /root/skype/siptosis.tgz
    cd /root


    []

  3. Starting xinit won't be a problem on the Aspire Revo. But, if xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for our purposes:

    Section "ServerLayout"
    Identifier "X.org Configured"
    Screen 0 "Screen0" 0 0
    EndSection

    Section "Device"
    Identifier "Card0"
    Driver "vesa"
    EndSection

    Section "Screen"
    Identifier "Screen0"
    Device "Card0"
    SubSection "Display"
    Viewport 0 0
    Depth 16
    Modes "800x600"
    EndSubSection
    SubSection "Display"
    Viewport 0 0
    Depth 16
    Modes "800x600"
    EndSubSection
    EndSection

    []

  4. If the test call fails with a bad audio message, go into Options, Sound Devices and reconfigure your Audio settings until you can place the test call successfully. Otherwise, none of the rest will work! []

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.

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 ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF

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*1061VoIP 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…

  1. passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. []
  2. status is the PIAF utility program that displays the current status of most major applications running on your server. []
  3. 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. []
  4. reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. []

It’s Orgasmatron 5.1: The Ultimate Asterisk Kitchen Sink

For those that want a turnkey Asterisk® VoIP PBX with every bell and whistle, today is your very lucky day. This tutorial will walk you through every step. In less than an hour, you'll have your very own, fully functional Asterisk PBX. No Linux skills are required for this setup. There's no charge for any outbound call made to any number in the U.S. or Canada. And inbound calls are free as well.

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

New Asterisk Security Model. Orgasmatron 5.1 has an all-new design which is intended to let you run an absolutely secure Asterisk PBX in your home from behind a secure firewall with NO INBOUND PORTS exposed to the Internet. So long as your router, Google Voice, and SIPgate passwords are secure, you can sleep like a baby. Today's Magic uses SIPgate as an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada. Remember that 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 trunk is used by Google Voice to call you back so the inbound SIPgate call is free. We handle all of 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 remember not to use extension 75 in Asterisk for your personal conferences!

Because we register your SIP connection with SIPgate permanently, there is no need to open the SIP or IAX Internet ports on your router. In short, your SIP connection with SIPgate works just as if you were using a browser behind a firewall. The return port will automatically be mapped by your NAT-based router. Hence, no security worries! We, of course, do 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.

Kitchen Sink Inventory. No kitchen is complete without an inventory. So, for those that are wondering what's included in the Orgasmatron 5.1 build, here's a feature list of the components you get in addition to the base PBX in a Flash build with Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. A2Billing, Cepstral, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.

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)

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 precautions to protect your phone bill. 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. 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. Configure SIPgate and Google Voice for Orgasmatron 5.1
2. Install the latest version of PBX in a Flash
3. Run the Orgasmatron 5.1 Installer
4. Enter your Google Voice and SIPgate credentials
5. Change existing passwords to secure your system
6. Configure a softphone or SIP telephone

Configuring SIPgate. A free SIPgate One residential phone number is a key component in today's project. This allows you to receive free incoming calls on your SIPgate number. Step #1 is to request an 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 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 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 SIPgate number and ask you to enter a confirmation code that's been provided. When your cellphone rings, answer it and punch in the number. Wait for confirmation. Then hang up.

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.

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

Installing PBX in a Flash. Now for the fun part. Here's a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta which 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. Download the 32-bit, PIAF 1.6 version from here, here, or here. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot your system 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.

Running the Orgasmatron 5.1 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.1 installer:

cd /root
wget http://pbxinaflash.net/orgasmo51.x
chmod +x orgasmo51.x
./orgasmo51.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!

Adding Your Credentials to PBX in a Flash. Now we're ready to insert your Google Voice credentials and SIPgate 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 RingBack DID (beginning with a 1). Don't get the 10-digit GV number mixed up with the 11-digit SIPgate RingBack DID, or nothing will work. 🙂

While logged into your server as root, 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!!

Next, run passwd-master and set your FreePBX passwords to something equally secure but different from your Linux root password.

Finally, type status and press Enter. Write down the IP address of your new server. You'll need it in the next step.

Configuring FreePBX. 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).

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 Changes, Apply Configuration Changes, Continue with Reload to save your settings.

Now click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate 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.

Whew! We recommend you reboot your server at this juncture just to be sure everything gets initialized correctly. Then all we need is a phone and you're all set.

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 your actual 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.

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 CID Superfecta in FreePBX. For tips, start here in the forums. Then 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. And, be sure to check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Finally, don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And be sure to add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

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. 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.

Upgrading Previous Orgasmatron V Installs. The question we hear over and over is "How do I upgrade from an existing Orgasmatron V install or from an existing Asterisk system?" The short answer is you can't. But there is some good news. For those with existing Orgasmatron V installs, we think we can fix your system so that it makes calls reliably. First, be sure your sipgate and gv-incoming settings match what is shown above in this article. Second, be sure you have configured a sipgate trunk with your proper sipgate credentials. Finally, log into your server as root and issue the following commands:
cd /root
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice-0.5*
cd pygooglevoice-0.5
python setup.py install
cd /etc/asterisk
sed -i 's|\${RINGBACK}|\${RINGBACK} 3|' extensions_custom.conf
asterisk -rx "dialplan reload"

Early Adopter WARNING. Current downloads are bug-free as best we can tell. But, for those that installed Orgasmatron 5.1 before 2:20 PM (EST) on Saturday, 2/27/2010, a couple of issues have arisen that need to be addressed. Please visit the following link to Orgasmatron 5.1 patches and apply those applicable to your particular situation. Without these patches, a security vulnerability may exist if you expose your server to web access from the Internet and a number of dialplan errors will cause unexpected behavior. It takes less than a minute to apply all of the patches! I'm reminded of the old Wild West adage: "You can always tell the pioneers by the arrows in their back."


Originally published: February 25, 2010


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

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.
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...

  1. No nastygrams! We know awesomest is not a 'real' word. Our spell-checker told us. 🙂 []
  2. For your final New Year's treat, be sure to watch the Cracker Rapper video! []

Introducing Phone Genie for Asterisk (Email Edition)

From Our Disney Cruise Family Scrapbook Almost two years ago, we introduced Phone Genie for Asterisk®. It let you reconfigure your Asterisk system remotely using your favorite web browser. This included the ability to set and adjust call forwarding, call waiting, and Do Not Disturb for any Asterisk extension. In addition, you could enter Asterisk CLI commands and execute a number of Linux system commands, all from the convenience of your web browser. Phone Genie for Asterisk remains one of the all-time favorite downloads of our readers.

Unfortunately, you don't always have access to a web browser when you're away from your Asterisk server. So today we introduce the perfect complement to the original Phone Genie with our new Email Edition. By following this quick tutorial, you can configure your Asterisk server to respond to any Asterisk CLI command which can be sent from almost any email client on the planet. And we'll perform all this magic with less than a dozen lines of bash scripting. Asterisk CLI commands have almost limitless possibilities. Use Phone Genie to check the status or change the functionality of just about any component on your server.

How It Works. The best way to explain how all of this works is to use a simple example. Let's assume you've left home and forgot to transfer your inbound calls for extension 701 to your cellphone. What we'll do is send a simple email message to a special user account on your Asterisk server that we've set up specifically to handle email directives for your server. Unlike most email addresses, we want this one to be unintuitive so strangers aren't sending messages to your server all the time. Let's assume the address is kxt1498@myserver.dyndns.org for this example. Using any email client, just address a message to that account. For the subject of the message, we'll use the following:

Asterisk: database put CF 701 6781234567

It doesn't really matter whether you include a message with the email. As long as the subject of the email is in the proper form, that's all that matters. The command above activates call forwarding for extension 701 and sends the calls to 6781234567. The command uses standard Asterisk CLI syntax.

On your Asterisk server, we'll have a simple bash script that runs every minute or two to check for new emails in the kxt1498 user's mailbox. If it finds a new message, it will parse the subject line, make certain there is a password match, and then send the command (unaltered) to the Asterisk Command Line Interface for processing. Here's an overview of all the CLI commands. The results of executing the command will be emailed to the address you've configured in the script. This works as both confirmation that your command has been executed and a security alert that your Asterisk system has been accessed using the Email Edition of Phone Genie. In the above example, you would receive an email at the address you've configured in the script with a subject of PhoneGenie. The body of the email would look like this:

Updated database successfully...database put CF 701 6781234567

Prerequisites. This software assumes you are using one of the Asterisk aggregations built on CentOS 5. We've tested it with PBX in a Flash. You'll also need an SMTP server (SendMail or Postfix) that is configured to send and receive emails to and from destinations on the Internet. You do not need a POP3 or IMAP mail server! We've tested this with Asterisk 1.4, but it should work fine with Asterisk 1.6 as well. FreePBX 2.5 or later is required for some functions.

Security Warning. Before we begin, let's pause for a moment to review the enormity of your problems if you do this wrong and to remind you that YOU ARE PROCEEDING AT YOUR OWN RISK. PBX in a Flash in particular is shipped with all outside access to your SMTP server blocked. We've obviously got to remove that layer of security for this software to function properly. But you need to be especially careful with SMTP servers because they can be used to relay SPAM to the entire world if you fiddle with settings that you don't understand. So... DON'T MAKE IMPROVEMENTS THAT AREN'T COVERED HERE UNLESS YOU KNOW WHAT YOU'RE DOING!

This software also gives certain email messages elevated privileges on your Asterisk server so that Asterisk itself can be reconfigured. If you compromise the email account name and password for this application, anybody worldwide can pretty much destroy the functionality of your server. In addition, calls to a certain extension could be rerouted to a very expensive destination on a cruise ship sailing around the world. If your dialplan permitted these calls and you had an account with automatic replenishment from a credit card or bank account, you've got a very expensive problem on your hands. That's one reason that reliable email notification of every Phone Genie transaction is critically important. If you're not getting timely notifications of each Phone Genie transaction, DO NOT USE THIS SOFTWARE until that problem is resolved!

Should you detect that your system has been compromised by receiving an email that indicates a command has been executed on your Asterisk server that you did not initiate, you should immediately disable or remove the script so that no further Phone Genie emails are processed on your server. Be sure to preserve any unprocessed Phone Genie emails for authorities as these may contain important information regarding the source of the emails. These email messages usually are deleted once Phone Genie completes execution of the associated Asterisk commands.

Overview. Here's the drill for today. First, we'll adjust both your hardware- based and IPtables firewalls to allow inbound email delivery to your Asterisk server. Second, we'll remove SendMail from your system and install and configure Postfix to handle the SMTP email chores. This will greatly simplify the security issues in locking down your server from unwanted emails. Depending upon your Internet service provider, installation of Postfix may break outbound email delivery from your server if your provider happens to block outbound traffic on port 25. We'll show you how to fix it. Third, we'll add a new user account on your Asterisk server that will be used exclusively to handle Phone Genie messages. Fourth, you're going to need a fully-qualified domain name for your Asterisk server so that email can be delivered reliably to your server. We'll walk you through getting this set up. Fifth, we'll install and configure the Phone Genie software and run some simple tests to make certain everything is working as it should. Sixth, we'll add the Phone Genie script as a cron job which will be run every couple of minutes to check for incoming Phone Genie emails. Finally, we'll review some of the Asterisk commands that can be executed using the Email Edition of Phone Genie for Asterisk.

Security Design. We've obviously given a great deal of thought to the security issues surrounding this application. The security model we've adopted works like this. First, for an email to get through to your Asterisk server, one and only one email address will work from the Internet. All other inbound email from the Internet will be rejected by Postfix. We strongly suggest you leave it that way. Your email address consists of the special username that we will create on your server plus a (hopefully new) fully-qualified domain name that points to your server. You are well advised to use and keep secret both a non-intuitive and complicated username AND a non-intuitive and complicated, fully-qualified domain name. Only this combination will let the email message through the Postfix filter! Using the correct username and a different FQDN that may also point to your server's correct IP address will nevertheless be rejected by Postfix. The third piece in the security model is the password. If you examine the sample Subject above, you will note that it begins with the word "Asterisk" followed by a colon, a space, and then the Asterisk CLI command. The word "Asterisk" is actually the password, and it can be changed to any password you like. So, if you change your password to FooBaR, then the subject of your message should look like this. Note that the colon followed by a space are also required!

FooBaR: database put CF 701 6781234567

Finally, it should be obvious but... DON'T SEND THESE EMAILS FROM AN UNTRUSTED CLIENT OR A PC IN A PUBLIC PLACE because your email message may get stored in a place that someone else could decipher how to access your server. If you wouldn't leave a $1000 bill beside the computer from which you're sending the email, don't send it! Otherwise, you may lose a good bit more than $1,000. To give you some idea of what's at risk with a compromised system, try sending the following email using your correct email address and password:

FooBaR: help

</sermon>

Firewall Configuration. For purposes of our example today, we're assuming that your Asterisk server is sitting behind a hardware-based firewall/router on a private subnet and that your Asterisk server includes a functioning software-based IPtables Linux firewall. This is the default PBX in a Flash setup that we always recommend. On your hardware-based firewall, you will need to redirect incoming TCP port 25 traffic to TCP port 25 on the private IP address of your Asterisk server. This change often requires a reboot of your firewall/router. Once that change is complete, log into your Asterisk server as root and edit /etc/sysconfig/iptables on PBX in a Flash systems. We need to add a new rule to IPtables which allows incoming TCP port 25 traffic through the firewall. Scroll to the bottom of the file and insert the following lines just above the COMMIT line:

# Allow inbound SMTP traffic on TCP port 25
-A INPUT -p tcp -m tcp --dport 25 -j ACCEPT

Save your additions to the file and then reload IPtables and your network:

service iptables stop
service iptables start
service network restart
service iptables status | grep "tcp dpt:25"

The last command should return an entry from IPtables showing TCP port 25 traffic is now being ACCEPTed into the server. If not, check your entries and repeat the process until this works.

Postfix Installation. Let's continue by removing SendMail from your server and installing Postfix. They both perform the same email functions, but the complexity of SendMail makes the likelihood of a configuration error too risky for us to sleep well. If you understand the intricacies of SendMail and feel comfortable implementing the security model we've described above, by all means, have at it. We'll be happy to share your results with the rest of our user community. In the meantime, here's the Postfix solution. While still logged into your server as root, issue the following commands to uninstall SendMail and install Postfix:

rpm -e --nodeps sendmail
yum -y install postfix

Choosing a Username and FQDN. Before we configure Postfix, you need to decide upon a user account name for your Asterisk server to manage Phone Genie messages. And you also need a fully-qualified domain name which points to the public IP address of your Asterisk server. As mentioned above, we strongly recommend that the username and FQDN be obscure and unguessable. For example, a combination of letters and numbers that don't spell words are good choices. Something like dlrpzh7b3@dhf34.nerdvittles.com will help you sleep well. If you don't have a static IP address and dedicated domain for your server that you can manage, then use an equally obscure FQDN from a provider such as dyndns.org. Something like dhf34.dyndns.org works. You then can configure your Asterisk server to automatically keep your dynamic IP address current. We're going to use these entries as examples below. Obviously, you should choose different entries!

To create the new user account on your server using whatever name you have chosen, here are the commands to issue while still logged into your server as root. Just substitute your chosen username for dlrpzh7b3 in both commands. Be sure to choose a secure password, too.

useradd dlrpzh7b3
passwd dlrpzh7b3

Configuring Postfix. Now let's get Postfix set up for maximum protection. First, move to postfix directory: cd /etc/postfix. Now edit main.cf: nano -w main.cf. Search for the inet_interfaces line in the file: Ctrl-W, inet_interfaces =. Add a hash mark to the beginning of each uncommented inet_interfaces line so that your entries look like this:

#inet_interfaces = $myhostname
#inet_interfaces = $myhostname, localhost
#inet_interfaces = localhost

Next, search for mydestination in the file: Ctrl-W,mydestination =. Comment out each of the lines except the one that looks like this:

mydestination = $myhostname, localhost.$mydomain, localhost

Now add the private IP address of your Asterisk server and your FQDN chosen above to the line so that it looks like this. Don't forget the commas and keep everything on one line.

mydestination = $myhostname, localhost.$mydomain, localhost, 192.168.0.118, dhf34.nerdvittles.com

Finally, move to the last line in the file and make it look like this, all on one line:

smtpd_recipient_restrictions = check_recipient_access hash:/etc/postfix/access, permit_mynetworks, reject_unauth_destination

Save your changes to the file: Ctrl-X, Y, then Enter. Now edit /etc/postfix/access. Move to the very bottom of the file and add two new lines with the following entries using the actual email address and FQDN you chose above instead of the examples. The first line tells Postfix to allow emails addressed to the specified email recipient. The next line tells Postfix to reject all other emails addressed to anyone at this domain. Other domains and public IP addressing are blocked by our mydestination entry above.

dlrpzh7b3@dhf34.nerdvittles.com OK
dhf34.nerdvittles.com REJECT recipient rejected

Save your changes to the file: Ctrl-X, Y, then Enter. Now issue the following two commands:

postmap /etc/postfix/access
service postfix restart

Testing Postfix. Now comes the important part. We need to make sure that outbound emails from your Asterisk server are delivered. And we need to make sure that incoming emails ONLY to the one email address you've designated are received and that all other emails from the Internet are rejected. We can't stress enough how important all three of these tests are. If your Postfix implementation doesn't pass all three, DO NOT PROCEED!

Testing outbound email with Postfix is easy. While logged into your server as root, issue the following command using a destination email address (instead of yourname@gmail.com) where you regularly receive emails:

echo "Hi there" | mail -s Test yourname@gmail.com

Count to 20 and refresh your email's Inbox. If the message is there, you've passed Test #1. If not, check your junk mail folder. If it's still not there, try another email address if you have one. Still no cigar? Then your Internet Service Provider is probably blocking email generated from downstream email servers. For tips on remedying the problem, see this message thread on the PBX in a Flash forums. You might also want to review the Postfix tutorial on dyndns.com. Here's another good tutorial on setting up a Gmail relay using Postfix. Then run the test again until you achieve success.

Testing inbound email to your designated email address is Test #2. Use a web client and send an email message to dlrpzh7b3@dhf34.nerdvittles.com substituting the actual email address you have chosen for your server. Count to 20, log into your server as root and type the following command to retrieve email for user dlrpzh7b3: mail -u dlrpzh7b3. The server should report that you have one new message. Type "d 1" and then "q" to delete the message and quit the mail app. If no email arrives, check the Inbox on your sending client to see if the message bounced and, if so, why. Check your email entries in /etc/postfix/access and /etc/postfix/main.cf for typos and review the steps in Configuring Postfix above. Then repeat the test until you successfully send a message to your designated email address.

Testing inbound email to an unauthorized email address on your Asterisk server is Test #3. For this test, we want to make sure that an email sent to the root account on your server fails. What you'll need for this test is the FQDN that was chosen above. Then, using a mail client, send an email message to root@dhf34.nerdvittles.com using your actual FQDN. Count to 20, log into your server as root, and type: mail. The message you sent should NOT be in the Inbox. Repeat the test by sending a message to root and dlrpzh7b3 @the actual IP address of your Asterisk server. These, too, should both fail. Once you get a passing grade on all three tests, we can move on. The hard part is behind you!

Installing Phone Genie. While logged into your server as root, issue the following commands:

cd /root
wget http://pbxinaflash.net/source/nv/phonegenie.tgz
tar zxvf phonegenie.tgz
rm phonegenie.tgz

Configuring Phone Genie. While still logged into your server as root, edit phonegenie.sh. You will note that there are 3 fields that need to be configured at the top of the file: user, pw, and notify. The user field is the designated user account name that will be used for incoming emails (dlrpzh7b3 in our example). The pw field is the word in every email Subject that precedes the colon, space, and Asterisk CLI command (Asterisk in our example). The notify field is a reliable email address where you regularly receive emails promptly. This is where the results of your Phone Genie email commands will be sent. Choose this email address wisely, as if your bank account depended upon it. It does! Once you have filled in the 3 fields (preserving the quotation marks around each entry), save the file with your changes.

Testing Phone Genie. Now we're ready to try everything out. Using an email client, send an email message to dlrpzh7b3@dhf34.nerdvittles.com (using your actual Phone Genie email name and FQDN). For the Subject, enter the following (substituting the password you created above for Asterisk)... Asterisk: help

After counting to 20, log into your Asterisk server as root and issue the following command:

/root/phonegenie.sh

You should see a display of all of the Asterisk CLI commands and within a minute or so, you should receive an email with the same information at the email address you entered into the notify field in phonegenie.sh in the previous step.

Installing Phone Genie as a Cron Job. Once you have tested several Phone Genie emails manually and you're satisfied that everything is working reliably, you can set up the Phone Genie shell script as a cron job. It should be set to execute every minute or every couple of minutes throughout the day and night. Edit /etc/crontab and insert the command shown below to have the script execute every 2 minutes:

*/2 * * * * root /root/phonegenie.sh > /dev/null

Sample Phone Genie Commands. In addition to all of the traditional Asterisk CLI commands, Phone Genie also supports a number of commands that are specific to FreePBX. These additional commands let you configure call forwarding, call waiting, do not disturb, system speed dials, and blacklist entries on your Asterisk server. For Asterisk CLI command syntax, consult voip-info.org. For FreePBX command syntax, see the listing below. Enjoy!

database put CF 302 8338116666 * Call Forwarding Enable
database del CF 302 * Call Forwarding Disable

database put CFB 302 8238221234 * Call Forwarding on Busy Enable
database del CFB 302 * Call Forwarding on Busy Disable

database put CFU 302 8038445689 * Call Forwarding Unavailable Enable
database del CFU 302 * Call Forwarding Unavailable Disable

database put CW 302 ENABLED * Call Waiting Enable
database del CW 302 * Call Waiting Disable

database put DND 302 YES * Do Not Disturb Enable
database del DND 302 * Do Not Disturb Disable

database put blacklist 6781234567 1 * Blacklist a number
database del blacklist 6781234567 * Remove blacklisted number

database put sysspeeddials 99 6781234567 * Set up Speed Dial 99
database del sysspeeddials 99 * Remove Speed Dial 99
(NOTE: Be sure you enable Feature Code *0 prefix in FreePBX!)

We wish all of you a very Merry Christmas!



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...

Free U.S. & Canada Calls: Google Voice Dialer for Windows

There now are a number of ways to make free calls to anyone in the U.S. and Canada using Google Voice without having to jump through the hoops of calling into your voicemail and having Google Voice call you back. There’s our Asterisk® implementation using pygooglevoice which lets you transparently place calls through Google Voice using any phone connected to your PBX in a Flash system. You also can set up a Sip Sorcery account and make free calls through that interface using a SIP phone. And now there’s Dogface05’s stand-alone Dialer for Windows that lets you place calls from the Windows command line in seconds. Because this is such a simple alternative, everyone should add it to their Windows toolkit. Here’s how.

Prerequisites. You’ll obviously need a Google Voice account. If you don’t have one, just register for an invite. Next, you’ll need a phone number to use for placing the outbound calls. And, finally, you’ll need to download and install Dogface05’s dialer on your Windows system.

Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of the phone you’ll be using to place calls and mark it as an Office phone. You’ll have to go through Google’s confirmation drill to successfully register the number with Google Voice. After the number is confirmed, be sure there’s a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to this number.

While you’re still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller’s Number. Finally, uncheck Do Not Disturb. Now click the Save Changes button.

Dialer Setup for Windows. From your Windows machine, open a browser and download the Google Voice dialer to your Desktop. Unzip the downloaded file and drag gvdial.exe to your \windows directory so that it’s in your path.

Placing a Call. Let’s first make sure everything is working properly. Open a command prompt window from the Windows Desktop and enter a dialing command using the following syntax:

gvdial username password destination ani [phonetype]

where:

  • username = your Google Voice email address
  • password = your Google Voice password
  • destination = 10-digit number of person to call
  • ani = your 10-digit phone number registered with Google Voice
  • phonetype = 3

The phonetype is actually optional and can be ignored unless you happen to be using a Gizmo number in which case it needs to be 7. Never enter the brackets. That merely signifies that the entry is optional.

Assuming your registered email address with Google Voice is joe@gmail.com, your password is secret, the number you wish to call is 6781234567, and your number is 4049876543, the dial string should look like this:

gvdial joe@gmail.com secret 6781234567 4049876543

Your phone should ring at this point, and Google Voice will complete the outbound call to 678-123-4567.

Creating Speed Dial Batch Files. Using Notepad, you now can create batch files for frequently dialed numbers. For example, the entry above could be saved in a batch file called joe.bat. Then simply create a desktop icon for Joe and link it to joe.bat. Double-click on the Joe icon whenever you wish to place a call to Joe. Here’s how the batch file might look:

echo off
cls
gvdial joe@gmail.com secret 6781234567 4049876543
echo Press ENTER key after the called party answers.
pause


Surfing the Google Wave. We’ve got a dozen Google Wave invites to give away during the next week. Just post a comment on any Nerd Vittles article, and we’ll put your name in the hat. Be sure to provide a Gmail address with your comment as this is required to take advantage of the Google Wave Preview. Here’s a sample for you to try once you have Google Wave credentials:



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…

CallerID Superfecta 2.2.2: International CNAM Directories

Unlike Willie Nelson, we’ve always nurtured our software projects hoping some would grow up to be cowboys. Thanks to Tony Shiffer, Jeremy Jacobs, and a whole host of new contributors (Patrick, Zorka, Nixi, UKstevef, and others), nothing even comes close to the success that CallerID Superfecta has enjoyed. What began as Asterisk® CallerID lookups from three sources with the original CallerID Trifecta four short years ago now provides an astonishing 27 CallerID lookup sources from around the world in the latest CallerID Superfecta 2.2.2. And the real beauty of this new beast is the utility permitting new lookup sources to be added without any further software modifications. That’s the tip of the iceberg.

For those that are new VoIP telephony, a brief history lesson will get you up to speed. When you make a phone call, telephone providers have traditionally passed your CallerID number to the receiving carrier while throwing your CallerID name in the bit bucket. The duty fell on the receiving carrier to look up your name in its directory and associate it with the CallerID number for delivery to the receiving telephone. If this sounds absolutely crazy, you’d be right. Who is in a better position to know the name of the calling party: the company initiating the call or the company receiving the call? Duh! That, of course, ignores the fact that the Bell System in particular was in monopoly preservation mode. Since they initially owned both companies, it really didn’t matter. Well, it does today and government regulators for some reason have completely missed this last vestige of the Good Old Boys telephone network. Does it make any sense that over half the phones in the world are mobile phones and you never know who’s calling? </rant>

When VoIP telephony came along, we obviously had to do something about that. Thus was born the CallerID Trifecta, an Asterisk/FreePBX tool allowing you to associate names with phone numbers using your computer and publicly available resources on the Internet. Version 2.2.1 introduced worldwide CallerID lookups. Unfortunately, there was a big gotcha. Some names in other countries use special non-ASCII characters, and delivery of those characters to some telephones sent the phones into the ozone. Not exactly our problem (HINT!), but folks did still want their phones to work. 🙄 CallerID Superfecta 2.2.2 fixes that by adjusting special characters to pure ASCII until the phone manufacturers catch up with the times.

The CallerID Superfecta Design. As originally implemented, CallerID Superfecta let you choose one or more lookup sources for incoming CallerID numbers. When an inbound call arrived, the sources were queried in a specified order, and the first source that provided a matching CallerID name won. The CNAM search result was returned to Asterisk for display on your phone instruments, and the lookup procedure ended. The problem with the original design was that newer and better lookup providers continued to appear so the hard-coded search order wasn’t necessarily ideal for every user or organization, and the providers kept changing formats to make lookups more challenging. In addition, when you receive a call from another country, it made little sense to look up that number in directories in which it obviously would not appear. CallerID Superfecta 2.2.2 fixes that with its new CallerID Schemes support. This lets you tailor CallerID lookups based upon dial strings just as you would do with FreePBX inbound and outbound routes.

What Else Is New? For openers there are a number of new lookup sources as well as some tweaks to older sources that stamped out a few (more) bugs from our previous, sloppy code. 🙂 You’ll also note there’s a new checkbox to Check for New Lookup Sources Online. This lets you easily import all the new lookup sources as they are added to the repository. The web user interface (UI) for FreePBX also has been reworked. You can prioritize the lookups in the order that best meets your needs, and you can tailor lookup sources to match specific CallerID number sequences. There’s also a debug function built directly into the web user interface. By entering a telephone number in the debug field and pressing the debug button on the form, the results from your selected lookup sources together with the latency of each enabled data source can be displayed on the form for you to review. This debug function greatly enhances troubleshooting while serving as a terrific tool to assist you in fine tuning which providers to actually enable and in what order. Providers who can’t be reached, or who perform too slowly, or who provide lousy results can be turned off completely or moved to the bottom of the search order. Finally, CallerID Superfecta 2.2.2 introduces prefix code hooks. This gives developers the ability to trigger an additional outside process when the Caller ID function is initiated. For example, this feature might be used in a call center to allow the system to automatically perform an ODBC query and bring up a customer record for use by a customer service representative.

Installing CallerID Superfecta 2.2.2 Installation or upgrade should be a snap on any of the FreePBX-based Asterisk aggregations including PBX in a Flash, trixbox, and Elastix. First, using a browser on your desktop PC, download CallerID Superfecta 2.2.2 from the Superfecta repository. Do not decompress the .tgz archive. Second, open FreePBX with your browser and choose Admin, Module Admin, Upload Module. Browse and select the superfecta-2.x.x.tgz module from your desktop and click the Upload button. When the upload completes, click local module administration. Scroll down and click CID Superfecta. Click the Install or Upgrade radio button depending upon whether you have previously installed the Superfecta FreePBX module. Click Process, Confirm, Return to install the new module. Reload the Asterisk dialplan when prompted.

Configuring CallerID Superfecta 2.2.2. There really are only two steps to bring CallerID Superfecta on line. First , we’ll configure the lookup sources and search order of the lookups. And then, for each inbound route on your Asterisk system, we’ll tell FreePBX to use CallerID Superfecta as the CallerID lookup source.



To configure CallerID Superfecta, click Admin, Setup, CID Superfecta in FreePBX. If you’re using PBX in a Flash or trixbox, be sure to insert the UserName maint and your FreePBX maint password in the fields provided under General Options. Then choose the Services you’d like to use for queries by clicking on the corresponding Enabled buttons. For those in the U.S., if you’re unfamiliar with previous versions of the product, we’d recommend you start with Addresses, White Pages, Yellow Pages, Any Who, and Telco Data. If you use the Asterisk Phonebook, AsteriDex, or SugarCRM, enable those options as well. Our rule of thumb in prioritizing the searches is to move your personal directories (Asterisk Phonebook, AsteriDex, and SugarCRM) to the top of the list. For the remaining choices, we recommend you start with the following search order: Addresses, White Pages, Yellow Pages, Any Who, and then Telco Data. Telco Data normally returns only the city and state of the caller, not the caller’s name. Who Called requires registration. Once you get everything squared away, click the Save button. Then key in a few known phone numbers in the Debug section of the form and click the Debug button to make sure everything is working as you expected. Take note of the retrieval times and the results and adjust the search order to meet your needs. Remember, the first match on a name using the search sources from top to bottom wins. The other search sources are never consulted for this number.

For additional configuration options and tips on configuring SugarCRM, see this thread on the PBX in a Flash Forum.

Once you’re satisfied with your lookup sources and the search order, the only remaining step is to designate CID Superfecta as the CallerID lookup source in your inbound routes. For each inbound route on which you want CallerID lookups performed, click Admin, Setup, Inbound Routes and choose the desired route from the column of routes on the right margin. Scroll to the CID Lookup Source section of the form and choose CID Superfecta from the dropdown box. Click Submit, Apply Config Changes, Continue to save your entry.

Adding Support for More Countries. So… here’s the challenge. We need to finish the rest of the world. If your country is not yet supported in the following list of directories, do us all a favor and post a comment with a good Internet source for reverse name directory lookups in your area of the globe. This means you can plug in a phone number and the directory will return the name of the person or business associated with that number. Even if you’re not a programmer, providing this information will assist greatly in making even more sources available to everyone down the road. Here’s the list as it stands today:

Adding Your Number to Directories. We know some of you are wondering how to get your VoIP number or Google Voice number added to the phone directories. It’s easy at least in the United States! Just go to www.listyourself.net and sign up. Enjoy!


Twitter Feeds on Nerd Vittles. If you glance over to the right column just above the Google Maps, you’ll see the current Twitter feed for @NerdUno. But did you know you also can read anyone else’s tweets or list from the same UI? Just scroll to the bottom of the frame and try one of these: voipusers (for the VoIP Users Conference feed) or voipusers/voip-users-conference (for recent tweets from all members of VUC). No need to type @. We’ll handle that keystroke for you. 🙂


Enhanced Google Maps. In case you haven’t noticed, we’ve added yet another Google Map to Nerd Vittles. Now, in addition to showing our location with Google Latitude, we also are displaying your location based upon your IP address. We’ll show you how to add something similar to any LAMP-based Linux system in coming weeks. It’s a powerful technology that has enormous potential. If you’re unfamiliar with Google Maps, click on the Hybrid and Satellite buttons and then check out the scaling and navigation options. Double-click to zoom. Incredible!


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.



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


 

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…