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 Footprint™ 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:
chmod +x 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:
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...
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.
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!
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.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
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Some Recent Nerd Vittles Articles of Interest...
- Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. [↩]
- Here are the actual commands in the skype-setup script if you'd prefer to execute them one at a time:
tar jxvf skype_static*
yum install xorg-x11-server-Xvfb
yum install qt4
yum install xterm
yum install libXScrnSaver.i386
chmod +x skype-start
cp skype-start skype/.
tar zxvf /root/skype/siptosis.tgz
- 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:
Identifier "X.org Configured"
Screen 0 "Screen0" 0 0
Viewport 0 0
Viewport 0 0
- 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! [↩]