Posts tagged: skype

7 Steps to Skytopia: Pain-Free Calls with Skype and Asterisk

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!

Originally published: Tuesday, November 1, 2011


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Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


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." []

Android 3 Deal of the Year: Acer Tab for Under $300

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!


Originally published: Tuesday, August 16, 2011




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.


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


Some Recent Nerd Vittles Articles of Interest…

How Good Can a $298 Android Tablet Be?

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!


Originally published: Wednesday, August 10, 2011




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.


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


Some Recent Nerd Vittles Articles of Interest…

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

Skype + Asterisk (still) = Beautiful Music + Free Phone Calls

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.


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!

Originally published: Thursday, May 26, 2011



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.




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.


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is 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! []

The Incredible PBX: Meet the New Kid on the Block

As much as we loved the moniker, the Orgasmatron build was in desperate need of a name change to more accurately describe its true heritage. We didn't look too far for just the right name. Meet The Incredible PBX!

Thanks to the Zero Internet Footprint™ design, it's the most secure Asterisk®-based PBX around. What this means is Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server.1 And you won't find a more full-featured Personal Branch Exchange™.

NEWS FLASH: Incredible PBX is now available for Asterisk 1.8! Go here.

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

The Incredible PBX is much more than just a name change. In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features tailored to meet the needs of the individual: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we'll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

The Incredible PBX Inventory. For those wondering what's included with The Incredible PBX, here's a feature list of components you get in addition to the base install of PBX in a Flash 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 provided scripts.

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

  • Broadband Internet connection
  • $200 PC3 on which to run The Incredible PBX or a Proxmox VM
  • dLink Router/Firewall. Low Cost: $35 WBR-2310  Best: DGL-4500
  • Free Google Voice account (Available in U.S. without an invite at this link)
  • Free SIPgateOne residential account (U.S. cell to get SMS invite) OR
  • Free IPkall IAX account (recommended for international users)

Installing The Incredible PBX. The installation process is simple and straight-forward. Just don't skip any steps. Here are the 5 Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls:

1. Install the latest version of PBX in a Flash
2. Download & run The Incredible PBX installer
3. Set up your two provider accounts
4. Configure a softphone or SIP telephone
5. Run the configure-gv credentials installer

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. We recommend you install the latest 32-bit version of PBX in a Flash. This new build works much 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 5.5 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 virtually identical payloads for all versions of PBX in a Flash.

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Or, if you are feeling really adventurous or if you have new, bleeding edge hardware, try our new 32-bit, PIAF 1.7 build which features CentOS 5.5. This new release is available from SourceForge or Google Docs. The MD5 checksum for the PIAF 1.7 build is 184cdb00142ccdd814b11de23fb00082.

Download the brand-new 32-bit PIAF 1.7.5.5. from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and type ksalt press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve! 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 choose PIAF-Silver 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
status

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. Write down the dynamic IP address assigned to your server after running the status command. You'll need it shortly.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source with The Incredible PBX. The version of Asterisk installed from our payload file is very stable.

Running The Incredible PBX Installer. Log into your server as root and issue the following commands to download and run The Incredible PBX installer:

cd /root
wget http://incrediblepbx.com/incrediblepbx.x
chmod +x incrediblepbx.x
./incrediblepbx.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!

Here's a short video demonstration of the Incredible PBX installer process:

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. If you are eligible, we strongly recommend a SIPgate One residential account for The Incredible PBX. However, you may elect to use an IPkall account as an alternative. Both are free; however, you cannot register The Incredible PBX to IPkall's servers so you'll need to punch a hole in your firewall to receive incoming calls from Google Voice and IPkall. This step is not necessary with SIPgate accounts since there is a permanent registered connection between The Incredible PBX and SIPgate's servers!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any 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 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 IPkall. If you're using IPkall as your intermediate provider, first log in to your hardware-based firewall/router and map UDP port 45694 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.

After your firewall is properly configured, 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.

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.

Configuring Google Voice. Google Voice no longer is by invitation only so, if you're in the U.S. or have a friend that is, head over to the Google Voice site and register. 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.

Google places a test call to your number so you'll have to delay it a bit for IPkall. If you're using SIPgate, go ahead and tell Google to place the test call which will be forwarded to your cellphone. Enter the two-digit code that's displayed when you're prompted to do so. With IPkall, wait until we finish running the credentials configurator below.

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.

If you're using SIPgate and you've confirmed your number, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to The Incredible PBX. We're ready to insert your credentials and SIPgate/IPkall information into The Incredible PBX. You'll need several 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 10-digit SIPgate or IPkall RingBack DID. You'll also need to reenter your passwd-master password which is used to configure CallerID Superfecta. Finally, you'll need to tell the configurator whether you're using a SIPgate or IPkall account. In the case of SIPgate, you'll also be prompted to enter your SIP ID and SIP password. These are NOT the same as your account credentials!!

Log back into your server as root and issue the following command to kick off the configurator: ./configure-gv.x. 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!! Once you've checked and double-checked your entries, press Enter and The Incredible PBX setup will be completed. You'll need to press Enter again when the script finishes to reboot your PBX. After the reboot, your system will have randomly-generated passwords for every extension and voicemail box that is preconfigured on your system. The DISA password also has been changed. We generate five-digit passwords. If you will sleep better with longer passwords, be our guest. They are easily reset using the FreePBX web interface described elsewhere in this article.

Finally, log back into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

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 Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

If you're using SIPgate as your provider with Google Voice, you're ready to place a test call. If you're using IPkall, we still need to verify your IPkall number with Google Voice. Return to Google Voice and tell it to place the test call to your IPkall number which you've already entered as your destination number. Your softphone will ring momentarily. Enter the two-digit code provided by Google Voice, and you're all set.

Incredible PBX 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!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

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.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. 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. We've secured all of your passwords except your root password and your passwd-master password, and we're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also 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.

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.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. Be safe!

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 - Incredible PBX 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 Incredible PBX adds a bunch of additional options. See if you can find all of them. Also check out Tweet2Dial which uses Twitter to make Google Voice calls, send free SMS messages, and manage your Incredible PBX.

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 as well. 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 perfect for off-site backups which we'll cover in a few weeks.

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.

Originally published: Monday, April 19, 2010

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Adding Skype to The Incredible PBX

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

Adding Multiple Google Voice Trunks to The Incredible PBX

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

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 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.

Coming Soon. We haven't forgotten. We'll cover setting up multiple Google Voice accounts for simultaneous calling on multiple channels very soon. And the new (free) Skype Gateway to Asterisk for The Incredible PBX is now available. The FreePBX components already are in place to support inbound and outbound calling via Skype. You can even try a test call to our Aspire One Revo today by dialing nerdvittles from your favorite Skype client. Beginning today, this article will be available on http://IncrediblePBX.com. Then Nerd Vittles will return to our (almost) weekly schedule of new articles. 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.


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


Some Recent Nerd Vittles Articles of Interest...

  1. Requires a SIPgate One account. []
  2. For Asterisk 1.6 or for 64-bit systems with Asterisk 1.4 or 1.6, use the Cepstral install procedures outlined in this Nerd Vittles article. []
  3. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. []
  4. 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. []

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 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:

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.


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID 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. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts 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 our 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 and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up 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. Any balance is refundable if you decide to discontinue service with Vitelity.


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! []