It’s been a long road, but we finally have a reliable Skype™ implementation for Asterisk® 11. Ironically, it uses components from ArchLinux™ as well as FreeSwitch™. But, hey, it works! It sounds great. And it lets you talk (for free) to over a half billion of your closest friends around the world. So today we present a Skype solution that works with a full-featured PBX. It can run on any Windows®, Mac®, or Linux® Desktop using the PIAF-Green™ Virtual Machine with Oracle’s free VirtualBox®. There’s more good news. It’s a 5-minute install once you’ve downloaded the PIAF-Green appliance. And the appliance includes a one-click installer for a Skype gateway. All you’ll need is a dedicated Skype account with your username and password. Before you can finish your cup of coffee, you’ll be up and running. For those that prefer to do it yourself, there’s a complete tutorial on the PBX in a Flash™ Forum. Our special thanks to @sukasem for solving the riddle.

PIAF-Green Virtual Machine: How It Works

We’ve written extensively about Oracle® VM VirtualBox and the PIAF-Green Virtual Machine. If all of this is new to you, start here for some background. For today, we’ve built a custom PIAF-Green appliance that’s designed to handle inbound and outbound calling using a dedicated Skype account. You can interconnect this appliance with other Asterisk servers, or you can use it as a standalone Asterisk server by adding extensions, trunks, and features just as you would with any other PIAF-Green server. By default, inbound calls to the Skype account on your server are routed to extension 701. That can be adjusted to meet your requirements. 11-digit outbound calls (e.g. 1-843-284-6844) are routed out through your Skype trunk which means you’ll need Skype Credit or a Skype Subscription to complete the calls. You can modify the outbound route SkypeOut to accommodate international calling, 10-digit calling, and free Skype-to-Skype calling worldwide if desired. We’ll cover the setup below. But, first, let’s get your PIAF-Green Virtual Machine up and running with FreeSwitch and Skype.

Installing the PIAF-Green Virtual Machine

Step #1 is to download the PIAF-Ast11-FS-Skype Open Virtualization Appliance (.ova) from SourceForge. This appliance includes a customized version of PIAF-Green with Asterisk 11 and FreePBX 2.11 as well as FreeSwitch and a Skype installer. Because Skype is proprietary, we cannot include it in the turnkey appliance, but the one-click installer will do all of the heavy lifting for you.

Step #2: Verify the checksums for the PIAF-Green 64-bit .ova appliance to be sure everything got downloaded properly. To check the MD5/SHA1 checksums in Windows, download and run Microsoft’s File Checksum Integrity Verifier.

For Mac or Linux desktops, open a Terminal window, change to the directory in which you downloaded the .ova file, and type the following commands:

md5 PIAF-Ast11-FS-Skype.ova (use md5sum for Linux)
openssl sha1 PIAF-Ast11-FS-Skype.ova


The correct MD5 checksum for the latest appliance release is 93b016298c18455184cfd596774c05ba. The correct SHA1 checksum for the latest release is 09b24ad03a99ecc4eeef58a5d123c3c03096a2e0.

The correct MD5 checksum for the original appliance was bfa436f4f4a1cf2dd5ac92c3fbb7c65f. The correct SHA1 checksum for the original appliance was 95228d4c2ed533092899da88721dbde8d20d8b73.

Step #3: Double-click on the downloaded .ova file which will begin the import process into VirtualBox. It only takes a couple minutes, and you only do it once. IMPORTANT: Be sure to check the Reinitialize the Mac address of all network cards box before clicking the Import button.

Once the import is finished, you’ll see the new PIAF virtual machine in the VM List of your VirtualBox Manager Window. You’ll need to make a single, one-time adjustment to the PIAF-Green Virtual Machine configuration to account for differences in network cards on different host machines.

Click on the PIAF-Ast11-FS-Skype Virtual Machine in the VM List. Then click Settings -> Network. For Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is ever necessary for your appliance. The rest is automagic.

Running the PIAF-Green Virtual Machine in VirtualBox

Once you’ve imported and configured the appliance, you’re ready to go. Highlight the Virtual Machine in the VM List on the VirtualBox Manager Window and click the Start button. The PIAF boot procedure with CentOS 6.4 will begin just as if you had installed PBX in a Flash on a standalone machine. You’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your PIAF VM.

Here’s what you need to know. To work in the PIAF Virtual Machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. Always shut down PIAF gracefully! Click in the VM window with your mouse, log in as root, and type: shutdown -h now.

Run PIAF-Green Virtual Machine behind a hardware-based firewall with no Internet port exposure!

To begin, position your mouse over the VM window and left-click. Once the PIAF VM has booted, log in as root with password as the password. Change your root password immediately by typing passwd at the command prompt. Now set up a secure maint password for FreePBX as well. Type passwd-master. If you’re not in the Eastern U.S. time zone, then you’ll want to adjust your timezone setting so that reminders and other time-sensitive events happen at the correct time. While logged into your server as root, issue the following command: /root/timezone-setup.

To access the FreePBX GUI, use a browser to log into your PIAF server by pointing to the IP address of the PIAF VM that’s displayed in the status window of the CLI. Click on the User button to display the Admin choices in the main PIAF Menu. Click on the FreePBX option to load the FreePBX GUI. You will be prompted for an Apache username and password. For the username, use maint. For the password, use whatever password you set up with passwd-master.

PIAF-Green Virtual Machine: Installing Skype with the One-Click Installer

Now the fun. First, be sure you’ve set up a Skype account that can be dedicated for exclusive use with PIAF-Green. Once you have your Skype username and password in hand, you’re ready to begin. While logged into your server as root, issue the following command: /root/skype-install.

Accept the license agreement and then you’ll be prompted for your Skype username (one word!) and password. Type carefully. Once you’ve checked your entries, press the Enter key to begin the install. Skype will be downloaded and installed after which FreeSwitch will be configured for use with PIAF-Green. The whole process takes about a minute.

PIAF-Green Virtual Machine: Taking Skype for a Test Drive

The easiest way to make sure everything is working is to assemble all of the needed components on your Windows or Mac desktop and then walk through an inbound and outbound call using Skype.

First, let’s get Skype and FreeSwitch running under PIAF-Green. From the command prompt, issue the following commands:

pkill skype
pkill Xvfb
sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh
/usr/local/freeswitch/bin/freeswitch


Second, switch back to your desktop PC or Mac and install the Skype client for your operating system. Make sure that you have a different Skype account that can be used for this testing. Once installed, run Skype on your desktop and log in with your different Skype credentials.

Third, while still on your desktop PC or Mac, download and install the Yate SIP softphone. Run the Yate client and choose Settings -> Accounts -> New. Plug in the IP address of your PIAF-Green Virtual Machine (type status in the Linux CLI if you don’t know it). For the extension, use 701. For the password use 1234secret. Make sure the client shows it registered. If not, check your IP address, extension, and password for typos.

Now let’s try a call to your PIAF-Green Virtual Machine using the Skype client on your desktop computer. Plug in the Skype name that you assigned to the virtual machine and click the dial button. Within a couple seconds, your Yate SIP softphone should start ringing. Click Answer on the phone and say a few words. Yes, there will be echo. You’re using both phones on the same desktop. Click Hangup. If you run the Asterisk CLI in the appliance window of VirtualBox, you can watch the entire process unfold with realtime alerts from both Asterisk and FreeSwitch.

As configured, you can use your Yate softphone to make outbound PSTN calls through Skype on the PIAF-Green Virtual Machine, but this requires you to have money in your Skype Credit account. If you have a positive balance, simply dial an 11-digit number, e.g. 1-843-284-6844. If you only want to make free calls to other people using Skype anywhere in the world, then you need to adjust your FreePBX dialplan slightly. Log into FreePBX as explained above and then make the following change to SkypeOut which is accessible by going to Connectivity -> Outbound Routes -> SkypeOut:

In the original version of the appliance, to place PSTN calls by dialing 10-digit numbers and to place calls to other Skype users by dialing *SkypeName from a SIP softphone, you needed to adjust the Dial Patterns for SkypeOut so that they looked like what is shown above. Then clicking Submit Changes and Apply Changes implemented the changes. With the current release, these features already are included.

PIAF-Green Virtual Machine: Auto-Starting Skype and FreeSwitch on Bootup

Once you’re sure everything is working as it should, you’ll find it more convenient to automatically start Skype and FreeSwitch whenever you boot up your PIAF-Green Virtual Machine. Just issue the following commands while logged into your server as root and then reboot to try it out:

echo "sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh" \
>> /etc/rc.d/rc.local
echo "ulimit -s 240" >> /etc/rc.d/rc.local
echo "/usr/local/freeswitch/bin/freeswitch &" >> /etc/rc.d/rc.local
reboot


PIAF-Green Virtual Machine: Skype Tips and Tricks

We’ve designed the one-click Skype installer so that you can rerun it at a later date if it becomes necessary to change your Skype credentials.

As a cautionary note, be advised that you cannot load Incredible PBX on this build without wiping out the Skype and FreeSwitch functionality, but we’re working on it.

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. The solution is to set up speed dial numbers for your Skype friends by adding a Speed Dial number to your FreePBX dialplan. Let’s set one up for Skype’s Call Testing Service to show how it’s done. Choose Applications -> Extensions -> Other (Custom) Device in FreePBX. Provide an Extension Number (8378 spells T-E-S-T) which will be the Speed Dial number. This could just as easily spell your friend’s name using a TouchTone phone. Next enter a Display Name for the extension, e.g. Test Skype. Then insert the following in the dial field. If you were setting up a custom extension for a friend, just replace echo123 with your friend’s actual Skype name using the syntax below. Save your entries and reload the dialplan when prompted.

SIP/echo123@127.0.0.1:5070

Now you can dial 8378 using your Yate softphone or any extension on your PBX to connect to the Skype Call Testing Service. Enjoy!

Tweet

Originally published: Friday, April 26, 2013

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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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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.

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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. As an added bonus for PIAF users, Vitelity is offering free porting of all domestic local and Toll Free DIDs through May 18. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity. 

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Some Recent Nerd Vittles Articles of Interest…

We’ve got lots of great news for you this week. So it’s hard to know where to start. Let’s begin on the hardware front. For frequent readers of Nerd Vittles, you know that we’ve been a fan of the Acer Aspire Revo since it was released almost two years ago. At that time, the market price was about $200. Today, NewEgg sells one for $350. What’s changed besides the price almost doubling? Well, not much if you’re looking for a home or SOHO VoIP server to handle your communications needs. You get a better version of Windows for the garbage can and a dual-core Atom processor. Neither one is really necessary for our purposes.

We try to stay away from do-it-yourself hardware projects, but this one was just too good to pass up. NewEgg has been featuring a couple of Foxconn barebones kits in the $100 range that require zero talent to build. Basically, you add a stick of RAM and a hard disk and Voilà, you’re done. We’ve been late to the solid-state drive (SSD) party so here was a golden opportunity to experiment. For about $100, you can purchase a 60 to 128 GB Type III SSD depending on the sale of the week. SSDs (not to be confused with STDs) provide an incredibly fast storage device. No moving parts, little heat, no noise. In short, a perfect VoIP platform for those needing a PBX with less than 50 extensions. Add $20 for a 4GB stick of notebook RAM, and you’ve got yourself an awesome little VoIP server with the footprint of about 3 packs of cigarettes (if you remember what those are). Buy a second one if you want redundancy. And, yes, a PIAF2™ app is coming soon to keep the units in sync. For now, check out this thread on the PIAF Forums for ordering details. You’ll also find detailed tips for getting WiFi functioning AND secure on the third page of the thread.

PIAF2: One Incredible Platform. So now that you’ve got VoIP hardware, what’s next? Here’s how we build up our systems today. Start by downloading the 32-bit PIAF2 ISO. Then make yourself a bootable thumb drive using a 1GB or larger flash drive. Our tutorial will show you how. Boot up your new server with the thumb and install PIAF2 with Asterisk® 1.8 and FreePBX® 2.9. Once you answer a few prompts, head out to lunch. Your server will be ready when you get back. Log into your server as root and install Incredible PBX™: install-incredpbx3. Want a fax server, too? Just run: install-incredfax2. And, if this is for personal use, then there’s now an easy option to add Skype as well: install-skype2. Want backups to a thumb drive? It’s finally ready!

Sounds simple? It is. But what about documentation? Well, we’ve got you covered there, too. For PBX in a Flash™ installation, it’s here. For Incredible PBX and Incredible Fax™, it’s here. For Skype, it’s here. And, for Incredible Backup™ and Restore (30-day beta), it’s here.

There are lots of choices in the VoIP space today. But Nobody Beats FREE.™ And the ease with which you can add every VoIP bell and whistle on the planet leaves PIAF2 with no rivals, period. The thanks, of course, goes to our compatriot, Tom King, who has worked tirelessly to make this simple enough for any Fifth Grader. Why not make a little contribution to the project once you’re up and running. You’ll be rewarded tenfold.

Originally published: Monday, March 5, 2012

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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.

---

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

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Some Recent Nerd Vittles Articles of Interest…

As you probably know, Digium® announced that Skype for Asterisk® would not be available for sale or activation after July 26, 2011. Here we are in November. So what to do? If you're looking for a commercial solution, you're S.O.L. But, if you have a non-commercial PBX for personal use¹, then keep reading. We'll walk you through, step-by-step, getting Skype integrated into your PBX in a Flash or Incredible PBX environment. It's easy, but it's a manual process. If you follow the steps below in order, you'll be up and running in about 15 minutes.

Prerequisites. For today's project, we're assuming you have an existing Incredible PBX server running CentOS 5.7. If not, here's our tutorial to get you up and running quickly. You'll also need a keyboard, mouse, and monitor. We strongly recommend a dedicated server such as an Atom-based PC. If you're using a virtual machine, then you'll need a sound card alternative. Try this: /sbin/modprobe snd-dummy.

UPDATE: We've revised this article a bit to accommodate PIAF2 with CentOS 6.2 and Incredible PBX 3. Keep in mind that Skype is a 32-bit application so we strongly recommend a 32-bit platform if reliability matters to you.

Step 1. For inbound Skype calling to work with other implementations including generic PBX in a Flash systems, you'll need to create a SIP URI for your Asterisk server: mothership@127.0.0.1. You do NOT need to expose the SIP port(s) of your Asterisk server to the Internet, and we strongly recommend that you don't! We've previously explained how to set up a SIP URI in this article. The Incredible PBX includes this SIP URI functionality out of the box.

Step 2. You'll also need Java 1.5. To see if it's included in your distribution, issue the following command: rpm -q jdk. If your particular Asterisk distribution doesn't have JAVA 1.5 or higher installed (rpm -q jdk), here's how to do it. Go to the Oracle Technology Network, sign up for a free Oracle web account and log in. While still logged in, accept the binary code license agreement, and click on this link to download jdk-6u12-linux-i586-rpm.bin. Then copy the file to /root on your Asterisk server. Make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin) and then run it. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Whew!

Step 3. You'll also obviously need a dedicated Skype account for your Asterisk server. If you don't have one to spare, download the Skype software for your Mac or Windows PC, and sign up for a free account. You can try out your account by calling our demo hotline: nerdvittles. Get this working on your Mac or PC before proceeding! Then be sure you log out and disable automatic logins on reboot, or you'll have a problem down the road with two machines trying to log in to a single Skype account.

Step 4. Now we're ready to install the remaining software components that your server will need to access Skype. Log into your Asterisk server as root and issue the following commands.

cd /root
mkdir skype
cd skype
wget http://download.skype.com/linux/skype_static-2.1.0.47.tar.bz2
tar jxvf skype_static*
yum -y install xorg-x11-server-Xvfb
yum -y install qt4
yum -y install xterm
yum -y install libXScrnSaver.i386 < == use this for CentOS 5.x
#yum -y install libXScrnSaver <== use this for CentOS 6.x
wget http://incrediblepbx.com/siptosis.tgz
cd ..
wget http://incrediblepbx.com/skype-start
chmod +x skype-start
cp skype-start skype/.
cd /
tar zxvf /root/skype/siptosis.tgz
cd /root/skype

If you'd prefer to avoid all the typing, you can issue the following commands to download a script that will do all the heavy lifting for you. This is for CentOS 5.x systems only:

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

For PIAF2 systems running CentOS 6.x, use this instead:

cd /root
wget http://incrediblepbx.com/skype-setup2
chmod +x skype-setup2
./skype-setup2

Step 5. Now there are a few steps to manually configure the software components so that the entire Skype startup process can be automated when your server boots in the future. To begin, you'll need to fire up X-Windows which puts your server in graphics mode. This is the only mode that Skype understands. While logged into your server as root, issue the following command: xinit

NOTE: 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 CentOS 5.x systems:

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

For PIAF2 users, some have reported issues on Atom machines with seeing a display at all after xinit loads. If this happens to you, don't panic. Simply log into your server from a PC or MAC using SSH. Then run: vncserver :1. Set a password for VNC, and then use a VNC client on your PC or Mac to access VNC at the IP address of your server on display port 1. Now you can continue with Step 6, below.

Step 6. Now we're ready to start up Skype, and get it properly configured. There are two important requirements. First, we want to make sure your credentials are saved for automatic login in the future. And second, we want to configure Skype to run in a minimized state each time it restarts. To begin, click in the white graphics window on your screen using your mouse and issue these commands:

cd /root/skype/skype_static-2.1.0.47
./skype

Click on the Accept button to accept the Skype license agreement. Once Skype loads, enter your Skype Name and Password. Before clicking on Sign In, be sure to check the Automatic Sign In box so that you'll be logged in automatically in the future. Once you're logged in, click on the blue S in the lower left corner of the window to access the Skype Main Menu. Then click Options. When the General tab displays, check the box which says Start Skype minimised in the system tray. Then click the Apply button. To test things out, click on the Sound Device tab and then Make a Test Call. Once you're sure everything is working, click the Close button. Now click on the blue S again and click Quit to shut down Skype.

Step 7. Now we're ready to integrate Skype into the SipToSis middleware so that Asterisk can communicate with Skype. Issue the following commands to start Skype in background mode and then start SipToSis. Be sure to write down the PID for Skype in case we need to kill the app if something goes wrong.

./skype &
cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype. Before clicking Yes, be sure to click the Checkbox to Remember This Selection for future connections! When you click Yes, you'll see the SipToSis CLI indicating that it's waiting for a Skype call.

If you've installed this on an Incredible PBX, Skype should now be functional. From another Skype account, just call the Skype Name that you used to set this up, and your Asterisk extensions should start ringing. To test outbound Skype calling, use an X-Lite softphone connected to an extension on your Asterisk server and dial *echo123 to access Skype's call testing service or *nerdvittles to access our demo.

All that remains is to configure your server to automatically start Skype and SipToSis whenever your system is restarted. Here's how. Press Ctrl-Alt-F2 to get a new login prompt on your server. Log in as root and issue the following command:

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

Now reboot your server and make sure everything is working.

Navigation Tips. Here are a few navigation tips for managing your Asterisk console on CentOS systems once Skype has been installed:

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 find the Skype PID: pidof skype. To kill Skype: kill pid#. 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.

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. One final note of caution. Do NOT expose UDP port 5070 to the Internet unless you first secure this port with a username and password to avoid Internet intruders using your gateway as a free Skype dialing platform! You do not need 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your Asterisk server so we recommend you keep it securely behind at least a hardware-based firewall.

FreePBX Design. For those not using Incredible PBX, here is the FreePBX setup that Incredible PBX uses and that we recommend. For outbound Skype calls, you have two choices.

1. To place a call to a regular phone number using SkypeOut (which costs you money), you'll simply dial 8 plus the area code and number. Our foreign friends will have to adjust their dialplans and /siptosis/SkypeOutDialingRules.props accordingly. Today's setup assumes 10-digit phone numbers!

2. To place a call to a Skype username using a softphone that supports SIP URI dialing such as X-Lite, you simply precede the Skype username with an asterisk, e.g. *echo123 will connect you to the Skype Call Testing Service or *nerdvittles will connect you to the Nerd Vittles Skype demo.

For incoming Skype calls, the default setup routes those calls to a SIP URI: mothership@127.0.0.1. Whether you point this URI to an extension, ring group, or IVR is up to you. In the default Incredible PBX build, the mothership URI is pointed to the Stealth AutoAttendant, an IVR that plays a welcoming message and then transfers the call to a ring group if no digit is pressed by the caller.

Configuring FreePBX. To put this setup in place, use a web browser to access FreePBX on your Asterisk server. You'll need to create a Custom Trunk and then an Outbound Route.

1. Choose Setup, Add Trunk, Add Custom Trunk. Fill in the form so that it looks like the following using your own CallerID number obviously:

When you're finished, click the Submit Changes button and then reload the dialplan when prompted.

2. Next choose Setup, Outbound Routes, Add Route. Fill in the form so that it looks like this:

When you're finished, click the Submit Changes button. Be sure to move this new OutSkype route to the top position in your Outbound Routes listing in the right margin! Then reload the dialplan when prompted.

3. If you're not using Incredible PBX, add a new DayNight Control 1 option while you're still in FreePBX. Just specify where you want calls routed for Day mode and Night mode. Then, here's the easy way to activate SIP URI support on your Asterisk/FreePBX server. Copy the [from-sip-external] context from the extensions.conf file in /etc/asterisk. Now copy the content into extensions_override_freepbx.conf. Be sure to preserve the context name in brackets! On a FreePBX 2.8 system, make it look like the following. The additions we're making are shown in bold below:

[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => mothership,1,Goto(app-daynight,1,1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)

Finally, reload your Asterisk dialplan, and we're finished with Asterisk and FreePBX setup:

asterisk -rx "dialplan reload"


Fedora Builds. For those using recent Fedora builds, these systems have a full implementation of X-Windows and KDE. Just start the system in mode5 (graphics mode), log in, run Skype in one window and start up SipToSis in a terminal window using the commands in Step 7 above. Authorize external use of Skype when prompted.

Where To Go From Here. Well, those are the basics. You now can make one outbound Skype call at a time from your Asterisk server, and you can receive an inbound Skype call on any Asterisk extension when Skype users call your regular Skype name. If you want multiple Skype account support, then you'll need to do some tweaking. What you'll need is the STS Trunk Builder toolkit which is free, but proprietary. Enjoy!

Tweet

Originally published: Tuesday, November 1, 2011

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Great News! Google Plus is available to everyone. Sign up here and circle us. Click these links to view the Asterisk feed or PBX in a Flash feed on Google+.

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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.

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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

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Some Recent Nerd Vittles Articles of Interest...

1. Excerpt from the Skype Terms of Service: "Subscriptions are for individual use only. Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means). Each subscription is to be used for your own personal communication purposes only, to make calls to another individual. The use of the subscription for commercial gain, such as calling numbers specifically to generate income for yourself or others by placing such calls, is not permitted. Unusual call patterns may be considered indicative of such use and may result in us terminating your subscription and blocking your User Account in accordance with paragraph 11.2." [↩]

We’ve never done back-to-back reviews of similar devices, but this week’s Target ad changes all of that. As you might expect, Acer has covered all of the bases with their entry into the dual-core Android 3 tablet sweepstakes. You may recall that we weren’t huge fans of the Motorola Xoom which promised a lot and delivered a boatload of vaporware. The Acer Iconia Tab A500 is not the Xoom. You not only get a microSD slot and Flash that actually work, but Acer has thrown in an HDMI port that can output 1080p video as well as a USB port that lets you connect your favorite USB devices including external hard disks. It performs this magic with an 8-10 hour battery life. And this week (only at Target) you can pick up this WiFi-only device for half the cost of the Motorola Xoom. In fact, after the gift card, it’s only a dollar more than the single-core Vizio Tablet that we reviewed last week.

Update: See the comments for equivalent deals just announced at NewEgg and CompUSA.

It’s difficult to describe the feel of the Acer Tab. Suffice it to say, it’s dimensions coupled with its sleek and sculpted design put it in the league with the iPad2 unlike the Xoom which felt chunky and clunky despite being an ounce lighter than the Acer.

As we mentioned last week, we don’t dive too deeply into the technical weeds in our reviews. If you want the technical assessment, check out this PC World review. What we prefer to evaluate is real-world usage of these devices. The Acer Tab has stunning performance. In addition to reading email and browsing the web, here’s the suite of applications which we think matter to most folks. We want to watch videos from YouTube and NetFlix. We want to stream music from Google Music and Spotify and read our Kindle books. We like to use Skype. And, yes, we also like Flash video support which works perfectly on the Acer tablet.

In addition to running Android 3, the Acer Tab boasts impressive hardware specs running a 1GHz Nvidia Tegra 250 dual-core processor with 1GB of RAM and 16GB of ROM. Add another 32GB easily with the microSD slot. The 10.1-inch tablet has a 1280-by-800 pixel display with a 16:10 aspect ratio that’s perfect for HD video content. We always prefer testing devices with real-world video content that we’ve shot so we can compare it to performance on other devices. Our Pawleys Island Parade video didn’t disappoint. It’s performance and color were as good or better on the Acer Tab than on Apple’s top-of-the-line 27″ iMac featuring a quad-core 2.93 GHz Core i7 processor with 8GB of RAM plus L2 and L3 cache. The same can be said with playback of complex Flash video. Netflix unfortunately is still a few weeks off although rooted Acer devices reportedly run it just fine.

On the music front, it doesn’t get much better than the Acer Tab. With Google Music or Spotify, the music world is your oyster. And the silver lining is that the Acer Tab is the one and only device that includes Dolby Mobile audio. Once you adjust the equalizer to match your taste in music, you’ll have sound quality to match that 20-pound boombox gathering dust in your basement.

In the communications department, Skype performed well although video calls are not yet supported. That’s unfortunate given the impressive specs on the Acer Tab’s two cameras. The Iconia Tab has a 5-megapixel rear-facing camera with flash in addition to a 2-megapixel front-facing camera for video conferencing. Finally, making and receiving free phone calls using either an Asterisk® server with CSipSimple or Google Voice using a $50 Obihai device and the free ObiON client for Android both worked great.

There’s only one word you’ll need to remember to take advantage of this Target deal: H-U-R-R-Y! This is a one-week only special, and Target offers no rainschecks. So call around until you find one. You won’t be sorry. And, as usual, Target offers a 90-day, no questions asked return policy which is second to none.

Google+ Invites Still Available. Need a Google+ invite? Drop us a note and include the word "Google+" and we’ll get one off to you. Come join the fun!

Our Favorite Android Apps. We’ve listed a few of our favorite apps below for those just getting started with Android. Enjoy!

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Tweet

Originally published: Tuesday, August 16, 2011

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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.

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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

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Some Recent Nerd Vittles Articles of Interest…

Pretty damn good in the case of the new 8″ Vizio Tablet. While it’s not going to take any speed awards when compared with the new Galaxy Tab 10.1, it does have a 1GHz processor with 512MB of RAM which delivers respectable performance with incredible battery life that rivals any iPad. Storage capacity is limited to 2GB, but you can add a 32GB microSD and meet any computing demands you may have. Currently the device is WiFi only.

As you might expect, Vizio knows a thing or two about televisions, and there’s a silver lining with the Vizio Tablet. Not only is an IR blaster included in the hardware, but you also get a giant TV remote that controls any combination of TVs, cable and satellite boxes, DVD and BluRay devices, and about 95% of the other video and audio components you will find on the planet. And it works as well or better than any of the pricey, high-end touchscreen (with a little screen) TV remotes that would easily put you in the Poor House. Say goodnight, Logitech. There’s also a front-facing 640×480 camera which easily suffices for video conferencing. No current video conferencing apps work, by the way, but it’s only been on the street for a week. The best news of all, you can pick one up at Costco or WalMart if you want one today. Or order it from Amazon if you prefer tax-free.

We don’t dive too deeply into the technical weeds in our reviews. If you want the technical assessment, check out this SlashGear review. What we prefer to evaluate is real-world usage of these devices. The Vizio Tablet passes with flying colors. In addition to reading email and browsing the web, here’s the suite of applications which we think matter to most folks. We want to watch videos from YouTube and NetFlix. We want to stream music from Google Music and Spotify and read our Kindle books. We like to use Skype. Sorry, Apple, we also like Flash video support which works perfectly on the Vizio Tablet even though it’s currently running Gingerbread.¹

Last, but not least, being a phone nerd, we obviously want to make and receive free phone calls using either an Asterisk® server with CSipSimple or Google Voice using a $50 Obihai device and the free ObiON client for Android. Both work great!

Of course, the usual Android favorites including Google+ with the exception of (the currently non-functioning) Huddle for video conferencing with up to 10 participants, Maps, Navigation, and Google Talk all work flawlessly. Gallery is perfectly synched with your Picasa photo collection which now can store unlimited photos at no cost through Google Plus. If you want to actually take professional photographs and make feature films, this isn’t the device for you. With the exception of Skype which is not yet available for this device (which was just released), everything else we’ve mentioned works great especially if you’re living on a budget. And, with the addition of Huddle in Google+, the absence of Skype support really doesn’t much matter any more. If you happen to need a Google+ invite, here’s a link compliments of Nerd Vittles. Finally, and pardon us for repeating, if you’re sick of wrestling with a half dozen remotes to watch television, this device is worth its weight in gold. You’ll be asking yourself why no one but Vizio was smart enough to think of it.

Vizio also had a better idea when it came to the Android user interface. As you can see in the photo above, there’s a top section where you can install your Favorite Apps. Immediately below that is your entire Applications collection. At the very bottom, there are five buttons which you can assign to your Must-Have Apps such as email, your web browser, the Google Market, Settings, and whatever else you happen to like.

Another nice touch that hasn’t been mentioned in many of the reviews is that Vizio has added a new keyboard option. If you remember the ergonomic keyboards that had the keys divided into two sections, Vizio has done much the same thing on the touchscreen which greatly improves typing for those that actually learned how. This keyboard, of course, can be toggled on and off depending upon your personal taste.

In conclusion, we think Vizio has hit a home run with this device. The price point, the feature set, the form factor, and the incredible battery life are just about perfect. We’ve listed a few of our favorite Android apps below to get you started. Enjoy!

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Tweet

Originally published: Wednesday, August 10, 2011

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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.

---

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

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Some Recent Nerd Vittles Articles of Interest…

1. Honeycomb has been promised for down the road. [↩]

It's been a disappointing week for Asterisk® with Digium®'s announcement¹ that Skype® for Asterisk will no longer be available for sale after July 26, 2011. While many suspected that Microsoft might have been behind the development, it turns out according to Skype² that this had been in the works for several months so that the company could better focus its attention on Skype for SIP. The problem with that argument, of course, is that at least for now you can't make outbound calls to Skype users with Skype for SIP unless the Skype users happen to be paying for a Skype to Go DID. So where do we go from here?

Well, the good news is that Asterisk 1.8.4.1 now appears to qualify as a stable release with fixes to several nasty bugs which caused some Cisco and Polycom phones to no longer connect. And Skype still produces a client for Linux. And Greg Dorfuss still produces SipToSis, an amazing product that lets Asterisk systems communicate with Skype... in both directions.

The problem has been that the most current release of Skype for Linux required GLIBCXX_3.4.9 which is not part of the CentOS 5.x distribution even though it is available in current releases of Fedora. What you never, ever want to do is mix and match components from one Linux distribution in another. They don't call it Dependency Hell without reason. But, as luck would have it, there's always a guru somewhere that's smart enough to get all the pieces working together.

For those using Incredible PBX, today's your lucky day. We've written a little script that takes all the components outlined above and makes them play nice at least on most Atom-based computers. You still need a sound card that's compatible with CentOS 5.6, but once you're past that hurdle, it's smooth sailing to integrate Skype into your existing system. We'll leave it to you to sort out the licensing issues which do not appear to be problematic if your system is solely for personal use. After all, Skype still produces a Linux client for use on Linux systems, and that's what Incredible PBX happens to run on.

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 Skype for Asterisk? For openers, today's 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.³

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 Acer Aspire Revo to host your Incredible PBX although most Atom-based PCs should work just fine. We don't recommend EEE PCs. With other hardware, your mileage may vary because CentOS 5.6 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.

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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 script. NOTE: We recommend you make a good backup of your system before you begin!

cd /root
wget http://incrediblepbx.com/skype/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: xinit⁴

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

skype.sh

Now you need to log in to Skype with your Skype name and password. In this latest version of Skype, we've noticed a quirk. Enter your password before you enter your username, or the system may not accept your username. If the screen appears frozen, press Ctrl-C and try it again. 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 Call⁵ 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 #2: 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.sh &

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 also 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.sh & 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!

Tweet

Originally published: Thursday, May 26, 2011

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Changes in PBX in a Flash Distribution. In light of the events outlined in our recent Nerd Vittles article and the issues with Asterisk 1.8.4, the PIAF Dev Team has made some changes in our distribution methodology. As many of you know, PBX in a Flash is the only distribution that compiles Asterisk from source code during the install. This has provided us enormous flexibility to distribute new releases with the latest Asterisk code. Unfortunately, Asterisk 1.8 is still a work in progress to put it charitably. We also feel some responsibility to insulate our users from show-stopping Asterisk releases. Going forward, the plan is to reserve the PIAF-Purple default install for the most stable version of Asterisk 1.8. As of June 1, Asterisk 1.8.4.1 is the new PIAF-Purple default install. Other versions of Asterisk 1.8 (newer and older) will be available through a new configuration utility which now is incorporated into the PIAF 1.7.5.6.2 ISO.

Here's how it works. Begin the install of a new PIAF system in the usual way by booting from your USB flash drive and pressing Enter to load the most current version of CentOS 5.6. When the CentOS install finishes, your system will reboot. Accept the license agreement, and choose the PIAF-Purple option to load the latest stable version of Asterisk 1.8. Or exit to the Linux CLI if you want a different version. Log into CentOS as root. Then issue a command like this: piafdl -p beta_1841 (loads Asterisk 1.8.4.1), piafdl -p 184 (loads Asterisk 1.8.4), piafdl -p 1833 (loads Asterisk 1.8.3.3), or piafdl -p 1832 (loads Asterisk 1.8.3.2). If there should ever be an outage on one of the PBX in a Flash mirrors, you can optionally choose a different mirror for the payload download by adding piafdl -c for the .com site, piafdl -d for the .org site, or piafdl -e for the .net site. Then add the payload switch, e.g. piafdl -c -p beta_1841.

Bottom Line: If you use the piafdl utility to choose a particular version of Asterisk 1.8, you are making a conscious decision to accept the consequences of your particular choice. We would have preferred implementation of a testing methodology at Digium before distribution of new Asterisk releases; however, that doesn't appear to be in the cards. So, as new Asterisk 1.8 releases hit the street, they will be made available through the piafdl utility until such time as our PIAF Pioneers independently establish their reliability.

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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.

---

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

---

Some Recent Nerd Vittles Articles of Interest...

1. "Skype for Asterisk will not be available for sale or activation after July 26, 2011. Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011. This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion. Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date. Thank you for your business." [↩]
2. Skype is a trademark of Skype, Inc. "Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium. By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand." [↩]
3. Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. [↩]
4. 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

   [↩]

5. 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! [↩]