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11/11/11: To Celebrate Nerd New Year’s, Please Welcome…
Just click on the image above to visit the site. Content is updated at least twice daily. As always, we welcome your content suggestions. Enjoy!
Originally published: Friday, November 11, 2011
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+.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
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
EndSectionSection "Device"
Identifier "Card0"
Driver "vesa"
EndSectionSection "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
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+.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
- 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.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
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.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
- Honeycomb has been promised for down the road. [↩]
Incredible PBX for Asterisk 1.8: Back from the Brink
Ever had one of those weeks? It was a wild ride these past 7 days with the introduction of Asterisk® 1.8.1 and some new Google Voice twists. And then there was our DNS provider Omnis.com that trashed name resolution for our primary domain, pbxinaflash.net, while claiming with a straight face that they didn't provide tech support for their own stupidity. Yikes! But where there's a will, there's always a way. And by Friday night, not only were all the issues sorted out but the Google Voice Gtalk interface in Asterisk 1.8 for free calling in the U.S. and Canada is now better than ever. Our special thanks to the Asterisk Dev Team and Tom King of the PIAF Dev Team for restoring peace in the valley. No more callback hoops for outbound calling. Free DIDs in most area codes. Instantaneous connections. Crystal clear calls. You can almost hear a pin drop. And Incredible PBX now brings you all this magic in a turnkey install that even a monkey could handle.
As if we needed another one, our other surprise last week was the Ebay appearance of a Nortel SIP Videophone labeled as a 1535, but it had no WiFi in either the hardware or in the particular software build. The merchant was as surprised as we were to discover the missing WiFi component and now has corrected the ad. But that won't make the WiFi reappear. For those of you still purchasing these phones (and they're worth it), read the fine print if WiFi or firmware upgrades matter to you. The Turkish models have neither. As anyone that tracks Ebay auctions will tell you, the law of supply and demand controls the price. These began in the $30 range and as recently as two weeks ago were selling for almost $80. They've now dropped back into the $50-$60 range. You're usually better off calling the merchant, and the more you buy, the better the price. Five Stars Telecom usually stocks the U.S. models. But ask to be sure.
So here's the drill today. Just download the brand new PBX in a Flash 1.7.5.5.4 ISO with the newly patched Asterisk 1.8 Purple payload. Then burn the ISO to a CD and boot your server from the PIAF CD. Choose the Purple Edition after CentOS installs which will load Asterisk 1.8.1 with FreePBX 2.8. Finally, run through the 5-minute install of Incredible PBX for Asterisk 1.8.1. In less than an hour, you'll have a turnkey, secure PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep your head spinning for months. Not only can you start enjoying free phone service immediately, but you'll have a robust PBX platform that will keep your eyes popping for months learning about all the features that would have cost you hundreds of thousands of dollars less than a decade ago. Did we mention that all of this telephone goodness is absolutely FREE!
Thanks to its Zero Internet Footprint™ design, The Incredible PBX also remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with minimal port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.
The Incredible PBX Inventory. For those that have never heard of 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 5.5, Asterisk 1.8, FreePBX 2.8, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Hamachi VPN, and Mondo Backups are just one command away and may be installed using some of the PBX in a Flash-provided scripts.
- Amazon S3 Cloud Computing
- AsteriDex
- Baseball Scores & Schedules
- CallerID Superfecta (FreePBX Module adds Names to CID Numbers)
- SIP Color Videophone Support
- CallWho for Asterisk
- Cepstral TTS for 32-bit, Asterisk 1.81 (/root/nv/install-cepstral)
- Preconfigured Email That Works with SendMail
- ENUMPLUS (Use FreePBX to configure)
- Extensions (16 preconfigured with random passwords)
- FAX with HylaFax & AvantFax (/root/incrediblefax.sh)
- Festival Server and Festival TTS for Asterisk (festival --server &)
- Flite TTS for Asterisk
- FONmail
- FreePBX Backups
- Gizmo5 (Free Calls to Gizmo5 users worldwide: 1747xxxxxxx*1089)
- Google Voice (preconfigured)
- Free Hamachi VPN (install-hamachi)
- Hotel-Style Wakeup Calls (FreePBX Module)
- ISN: FreeNum SIP Calling from Any Phone
- MeetMe Conference Bridge (just dial C-O-N-F)
- Mondo Full System Backups (install-diskbackup)
- Incremental Daily Backups (install-dailybackup)
- Munin Reports (install-munin)
- NewsClips from Yahoo
- ODBC Database Support
- New PBX in a Flash Registry (show-registry)
- PogoPlug Cloud Computing
- Reminders by Phone and Web
- SAMBA Windows Networking (setup-samba)
- SIP URI Outbound Calling (call any SIP URI worldwide for free)
- Free Skype Inbound & Outbound Calling (for personal use)
- Stealth AutoAttendant
- TeleYapper
- TFTP Server (setup-tftp)
- Tide Reports with xTide
- Trunk Lister Script (/root/nv/trunks.sh)
- Trunks (Vitelity, Gtalk, SIPgate, IPkall, and ENUM)
- Twitter Interface (Make Free Calls and Send SMS Messages)
- Weather by Airport Code
- Weather by ZIP Code
- Worldwide Weather
Prerequisites. Here's what we recommend to get started properly:
- Broadband Internet connection
- $250 Dual-Core Atom PC2 on which to run The Incredible PBX
- dLink Router/Firewall. Low Cost: $35 WBR-2310 Best: DGL-4500
- Dedicated Google Voice account (not your main Gmail account!)
Installing The Incredible PBX. The installation process is simple and straight-forward. Here are the 5 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:
1. Install PBX in a Flash 1.7.5.5.4 Purple Edition
2. Download & run The Incredible PBX 1.8 installer
3. Run passwd-master on your PIAF server
4. Map UDP 5222 on firewall to PIAF server
5. Configure a softphone or SIP telephone
Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. To use Incredible PBX for Asterisk 1.8, we recommend the very latest 32-bit version of PBX in a Flash 1.7.5.5.4.3 If you installed it last week, that's not new enough. The ISO hasn't changed, but the Purple payload is radically different since this morning! Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.5 operating system. That hasn't changed. But, once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. To get the patched version of Asterisk 1.8.1, use today's new 1.7.5.5.4 ISO. Choose the new Purple Payload, and our special Asterisk 1.8 patched release and all of the Google Voice goodies will be configured automatically. And you won't have to worry about the CDR crashing your new server either.
You can download the 32-bit PIAF 1.7.5.5.4 from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin.
WARNING #1: 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.
WARNING #2: The PIAF Dev Team currently classifies PIAF-Purple and Asterisk 1.8 as E-X-P-E-R-I-M-E-N-T-A-L. Remember the Pioneers! If you have a low threshold for pain, if you depend upon your PBX to actually make and receive phone calls, or if you understand the WAF and prefer sleeping with both eyes closed, abort this install now and choose PIAF-Gold, PIAF-Silver, or PIAF-Bronze. Otherwise, enjoy the ride!
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'. 🙂 If your system still won't boot, then you have an incompatible drive controller.
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 PIAF-Purple option. Have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. 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 IP address of your new PIAF server. You'll need it to configure your hardware-based firewall in a minute.
NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs and update-fixes for you. You still should set your FreePBX passwords by running passwd-master after The Incredible PBX installer finishes!
Configuring Google Voice. You'll need a dedicated Google Voice account to support The Incredible PBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So why take the chance. Keep this account a secret!
We've tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with The Incredible PBX. 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. If you're living on another continent, see MisterQ's posting for some tips on getting set up.
You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work... in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...
IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That's the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don't see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.
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.
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/incrediblepbx18.x
chmod +x incrediblepbx18.x
./incrediblepbx18.x
passwd-master
If you've installed the previous version of The Incredible PBX, you'll recall that there was a two-step install process after configuring another trunk with either SIPgate or IPkall. That's now a thing of the past. All you need to do after The Incredible PBX script completes is run passwd-master to set up your master password for FreePBX.
When The Incredible PBX install begins, you'll be prompted for the following:
Google Voice Account Name
Google Voice Password
Google Voice 10-digit Phone Number
Gmail Notification Address
FreePBX maint Password
The Google Voice Account Name is the Gmail address for your new dedicated account, e.g. joeschmo@gmail.com. Don't forget @gmail.com! The Google Voice Password is the password for this dedicated account. The Google Voice Phone Number is the 10-digit DID for this dedicated account. We need this if we ever need to go back to the return call methodology for outbound calling. For now, it's not necessary. But who knows what the future holds. 🙄 The Gmail Notification Address is the email address where you wish to receive alerts when incoming and outgoing Google Voice calls are placed using The Incredible PBX. And your FreePBX maint Password is the password you'll use to access FreePBX. You'll actually set it by running passwd-master after The Incredible PBX completes. We need this password to properly configure the CallerID Superfecta for you. By the way, none of this confidential information ever leaves your machine... just in case you were wondering.
Now have another 5-minute cup of coffee, and consider a modest donation to Nerd Vittles... for all of our hard work. 😉 You'll find a link at the top of the page. While you're waiting (and so you don't forget), go ahead and configure your hardware-based firewall to support Google Voice. See the next section for what's required. Without completing this firewall configuration step, no calls will work! When the installer finishes, READ THE SCREEN just for grins.
Here's a short video demonstration of the original Incredible PBX installer process. It still works just about the same way except there's no longer a second step to get things working.
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!
Firewall Configuration. We hope you've taken our advice and installed a hardware-based firewall in front of The Incredible PBX. It's your phone bill. You'll need to make one adjustment on the firewall. Map UDP 5222 traffic to the internal IP address of The Incredible PBX. This is the port that Google Voice uses for phone calls and Google chat. You can decipher the IP address of your server by logging into the server as root and typing status.
Logging in to FreePBX. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Admin tab and choose FreePBX. When prompted for a username, it's maint. When prompted for the password, it's whatever you set up as your maint password when you installed Incredible PBX. If you forget it, you can always reset it by logging into your server as root and running passwd-master.
Extension Password Discovery. If you're too lazy to look up your extension 701 password using the FreePBX GUI, you can log 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 or color videophone 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 such as the $50 Nortel color videophone we've recommended above. You'll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you're like us, we want to make damn sure this stuff works before you shell out any money. So, 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.
Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. If not, make certain you are not logged into Google Chat on a Gmail account with these same credentials. 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. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) 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.
Adding Multiple Google Voice Trunks. Thanks to rentpbx on our forums, adding support for multiple Google Voice trunks is now a five-minute operation. Once you have your initial setup running smoothly, hop on over to the forums and check out this Incredible solution.
Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly 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 trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.
Using ENUMPlus. Another terrific money-saving tool is ENUM. Your system comes with ENUMPlus installed. The advantage of ENUM is that numbers registered with any of the ENUM services such as e164.org can be called via SIP for free. You can read all about it in this Nerd Vittles' article. To activate ENUMPlus, you'll need to register and obtain an API Key at enumplus.org. It's free! Sign up, log in, and click on the Account tab to get your API key. Once you have your key, copy it to your clipboard and open FreePBX with your browser. Then choose SetUp, ENUMPlus and paste in your API Key. Save your entry, and you're all set. After entering your key, all outbound calls will be checked for a free ENUM calling path first before using other outbound trunks.
Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it's merely a greeting. To those "in the know," it's actually an autoattendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It's quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It's your phone bill. 😉
Configuring Email. You're going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Setup, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Setup, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.
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. We also recommend you follow @NerdUno on Twitter. We'll keep you entertained and provide immediate notification of security problems that we hear about. Be safe!
This latest version of Incredible PBX locks down your server to private networks and existing, registered Asterisk devices. Should you need to enable additional IP addresses for other devices or providers at a later date, simply add the new IP addresses to /etc/firewall.whitelist and then rerun /root/firewall-whitelist.sh. For additional background, read this article.
Enabling Google Voicemail. Some have requested a way to retain Google's voicemail system for unanswered calls in lieu of using Asterisk voicemail. The advantage is that Google offers a free transcription service for voicemail messages. To activate this, you'll need to edit the [googlein] context in extensions_custom.conf in /etc/asterisk. Just modify the last four lines in the context so that they look like this and then restart Asterisk: amportal restart
;exten => s,n(regcall),Answer
;exten => s,n,SendDTMF(1)
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => s,n,Goto(from-trunk,gv-incoming,1)
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 - PIAF Support Conference Bridge (Conf#: 1061)
- 882*1061 - VoIP Users Conference every Friday at Noon (EST)
PBX in a Flash SQLite Registry. Last, but not least, we want to introduce you to the new PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here's the complete list of options:
- A - Write the contents of the registry to registry.txt
- B - PBX in a Flash install details
- C - Extra programs install details
- D - Update-fixes status and details
- E - RPM install details
- F - FreePBX modules install details
- G - Incredible PBX install details
- Q - Quit this program
And here's a sample from an install we just completed. We'll have more details and additional utilities for your use in coming weeks.
Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. Please bear with us. To Mark Spencer, Malcolm Davenport, and the rest of the Asterisk development team, thanks for a much improved Asterisk. To Philippe Sultan and his co-developers, thank you for getting the final kinks out of Jabber with Asterisk. To Philippe Lindheimer & Co., thanks for FreePBX 2.8 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really turned Asterisk 1.8.1 into something special. No one will quite understand what an endeavor that was until they try it themselves. And, finally, to our legion of beta testers, THANK YOU! We've implemented almost all of your suggestions.
Additional Goodies. 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. The 32-bit install-cepstral script does just what it says. With Allison's Cepstral voice, you'll have the best TTS implementation for Asterisk available. ipscan is a little shell script that will tell you every working IP device on your LAN. trunks.sh tells you all of the Asterisk trunks configured on your system. purgeCIDcache.sh will clean out the CallerID cache in the Asterisk database. convert2gsm.sh shows you how to convert a .wav file to .gsm. munin.pbx will install Munin on your system while awstats.pbx installs AWstats. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. All the other scripts and apps in /root/nv already have been installed for you so don't install them again.
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 and is included as one of the backup options in the PBX in a Flash backup utilities.
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. Enjoy!
Originally published: Monday, December 13, 2010
Quirks & Bugs. Well, there aren't any that we know of. But we'll keep a running list here so you can check back from time to time if you don't participate in the PBX in a Flash Forums.
VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions
Safely Interconnecting Asterisk Servers for Free Calling
Adding Skype to The Incredible PBX
Adding Incredible Backup... and Restore to The Incredible PBX
Adding Remotes, Preserving Security with The Incredible PBX
Remote Phone Meets Travelin' Man with The Incredible PBX
Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won't have to wait long for an answer to your questions.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
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Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
- For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. [↩]
- 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. You also can run The Incredible PBX on a virtual machine such as the free Proxmox server. Another less costly (but untested) option might be this Shuttle from NewEgg: $185 with free shipping. Use Promo Code: EMCYTZT220 [↩]
- HINT: Version 1.7.5.5.3 ISO also works just fine. [↩]
Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk
It’s been an exciting couple of weeks watching the overwhelmingly positive response to our release of Orgasmatron 5.1. With this version, we introduced a new Asterisk® security model that took into account the ever-increasing security risks posed by exposing web and telephony servers to direct Internet access. The bottom line is this. If your telecom requirements still can be accomplished by placing a server securely behind a $35 hardware-based Internet firewall with no Internet exposure, then it makes absolutely no sense to dangle such a tempting target in front of the world’s most nefarious creeps.
News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!
Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11
Our experience suggests that the only trade off with this new approach is the inability to receive anonymous SIP calls… a small price to pay considering the potential financial and computer risks involved. You still can place outbound VoIP calls as well as placing and receiving calls using any of the phone numbers registered on your new PBX in a Flash server. And, thanks to Google Voice, SIPgate, and IPkall, all inbound calls are free, and all outbound calls to numbers in the U.S. and Canada are free as well.
If a SIP URI and your own Freenum/ISN number are simply features you can’t live without, sign up for a voip.ms IAX account, and you’ll get a SIP URI for free. Inbound SIP URI and Freenum/ISN calls will set you back $1 for every 1,000 minutes billed in 6 second increments.
Or you can sign up for a free IP Freedom CallCentric account and configure a new SIP trunk in FreePBX by following these directions. Once configured, your new server SIP URI will be 1777xxxxxxx@in.callcentric.com where xxxxxxx is your assigned 7-digit CallCentric number.
Keep in mind that a new security vulnerability has been found with either Asterisk or FreePBX almost monthly. The chart below tells you why. With virtually limitless attack surfaces because of the number of interrelated components in CentOS, Asterisk, and FreePBX comes enormous and recurring potential for remote compromise of these systems. Rather than play this cat-and-mouse security game with the underworld, the Orgasmatron design changes the paradigm. It lets you use any (secure or insecure) version of Asterisk and FreePBX without worrying about any outside attacks. Do passwords on your new server matter? Not really… unless there is someone inside your firewall that you don’t trust. 🙄 Are we going to secure them anyway? Absolutely. But instead of the constant worry over new security vulnerabilities, Orgasmatron 5.2 lets you enjoy exploring the world of Asterisk and VoIP telephony with an incredibly rich feature set that you won’t find anywhere else, period! We’ll resist making any other device analogies, but the idea here is to protect the good guy (you!) while keeping the bad guys out. No penetration. No worries. Simple as that.
In our former life working for a living, we actually procured and managed multimillion dollar PBXs as part of our "other duties as assigned." Without qualification, we can tell you that the feature set that Orgasmatron 5.2 brings to the table for free runs circles around anything you could buy (then or now) in the commercial marketplace. And, at one time or another, we purchased every Nortel feature good money could buy. There’s one other difference. Orgasmatron 5.2 runs swimmingly on a $200 Atom-based PC that you can purchase at any Best Buy as well as hundreds of other stores including Amazon, NewEgg, and Buy.com. We paid more than $200 to provision an additional extension on our Nortel switch! You, of course, can add as many extensions as you like. De nada.
So, why a new version of Orgasmatron in only a few weeks? Well, it’s not security-related. In fact, there is nothing wrong with continuing on with Orgasmatron 5.1. Unfortunately, it relied exclusively upon SIPgate to make free Google Voice calls in the U.S. and Canada. And SIPgate required an invite using an SMS message from a U.S.-based cellphone. That pretty well knocked out all of our friends living outside the United States. Today’s version fixes that by letting anyone sign up for a free IPkall phone number in Washington state. All you need is a valid email address. The setup process is a bit more complex because IPkall doesn’t support registered connections to their servers. But we’ll walk you through the additional steps and, once completed, your server will be just as secure as the SIPgate approach we set up with Orgasmatron 5.1. And few, if any, Linux skills are required to set up or manage Orgasmatron 5.2. As we’ve noted previously, if you can handle slice and bake cookies, you’ve got the necessary skillset! Be aware this is about a one-hour project, and you need to track through the article carefully, or the entire house of cards comes down.
New Asterisk Security Model. Orgasmatron 5.2 maintains our design goal of running an absolutely secure Asterisk PBX from behind a hardware-based firewall with either NO INBOUND PORTS exposed to the Internet with SIPgate or an IP-address-restricted IAX port for IPkall. Don’t defeat this security mechanism by exposing additional ports on your PBX in a Flash server to Internet access. And choose your NAT-based firewall/router carefully. All of these devices are not created equally. Not only do some perform better than others, but certain models are notoriously bad at handling NAT-based routing tasks, a critical requirement in the Asterisk VoIP environment. In almost every case of problems with one-way audio, the real culprit can be traced back to a crappy router. For $35, you really can’t go wrong with the dLink WBR-2310. If you want traffic shaping functionality as well, take a look at dLink’s Gaming Router, our personal favorite.
As long as your router, Google Voice, SIPgate, and IPkall passwords are secure, you can sleep like a baby. We use an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada because Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you’re calling is connected. The SIPgate or IPkall trunk is used by Google Voice to call you back so the inbound call is always free. We handle the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just refrain from using extension 75 in Asterisk for personal conferencing!
The choice is yours. You can use SIPgate with no incoming ports exposed to your server from the Internet. Or you can use IPkall and map UDP port 4569 (IAX2) on your hardware-based firewall to the internal IP address of your new PBX in a Flash server. Even with the IPkall setup, we’ve locked down IPtables (our Linux firewall) to restrict IAX access to several specific IP addresses so your server remains absolutely secure. We’ve also included support for FonicaTec’s IAX offering for those that want a backup IAX provider. We’ll have much more to say about IPtables in coming weeks.
If you’ve already installed Orgasmatron 5.1 and it’s working for you, do you need to upgrade? NO. With the exception of the new IAX support for IPkall, the code in Orgasmatron 5.2 is identical.
We, of course, continue to recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.
Swiss Army Knife Inventory. There’s no need for a Swiss Army Knife if you don’t know what all the blades are for. So, for those that are wondering what’s included in the Orgasmatron 5.2 build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.
- A2Billing (/root/nv/install-a2billing)
- Amazon S3 Cloud Computing
- AsteriDex
- CallerID Superfecta (FreePBX Module)
- CallWho for Asterisk
- Cepstral TTS (/root/nv/install-cepstral.sh)
- Preconfigured Email That Works with SendMail
- Extensions (16 preconfigured)
- Fax Module using nvFax
- FONmail
- FreePBX Backups
- Gizmo5 (Free Calls to Gizmo5 users worldwide: 1747xxxxxxx*1089)
- Google Voice (preconfigured)
- Hamachi VPN (/root/nv/install-hamachi.x)
- Hotel-Style Wakeup Calls (FreePBX Module)
- ISN: FreeNum SIP Calling from Any Phone
- MeetMe Conference Bridge (just dial C-O-N-F)
- Mondo Full System Backups (/root/nv/install-diskbackup.x)
- NewsClips from Yahoo
- ODBC Database Support
- PogoPlug Cloud Computing
- Reminders by Phone and Web
- SIP URI Outbound Calling (call any SIP URI worldwide for free)
- TeleYapper
- Tide Reports with xTide
- Trunk Lister Script (/root/nv/trunks.sh)
- Trunks (Vitelity, Fonica, SIPgate, IPkall, and ENUM)
- Twitter Interface (Make Free Calls and Send SMS Messages)
- Weather by Airport Code
- Weather by ZIP Code
- Worldwide Weather
- Zaptel Updater (/root/nv/zaptel-update.sh)
Prerequisites. Here’s what you’ll need to get started:
- Broadband Internet connection
- Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
- $200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
- Free Google Voice account (HINT: Under $2 on eBay)
- Free SIPgateOne residential account (Use cell to get SMS invite) OR
- Free IPkall IAX account
Learn First. Install Second. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today’s VoIP world. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.
Today’s Drill. The installation process is straight-forward, but a little different than the Orgasmo 5.1 scenario because of the need to accommodate IPkall. Just don’t skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:
1. Install the latest version of PBX in a Flash
2. Run the Orgasmatron 5.2 Installer
3. Configure a softphone or SIP telephone
4. Configure Providers for Orgasmatron 5.2
5. Enter your Google Voice and SIPgate/IPkall credentials
6. Change existing passwords to secure your system
Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta on a new Atom-based PC. This beta is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use the identical payload for versions 1.3, 1.4, 1.5, and 1.6 of PBX in a Flash. The beta label simply means we haven’t had time to sufficiently test CentOS. But this is not a Microsoft-style beta so fear not!
Download the 32-bit, PIAF 1.6 version from SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot from the installation CD and type ksalt to begin.
WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! Press Ctrl-C to cancel the install.
On some systems you may get a notice that CentOS can’t find the kickstart file. Just tab to OK and press Enter. Don’t change the name or location of the kickstart file! This will get you going. Think of it as a CentOS ‘feature’. 🙂
At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:
update-scripts
update-fixes
When prompted, change the ARI password to something really obscure. You’re never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.
NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source on the Orgasmatron builds because of parking lot issues in the latest releases of Asterisk.
Running the Orgasmatron 5.2 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.2 installer:
cd /root
wget http://pbxinaflash.net/orgasmo52.x
chmod +x orgasmo52.x
./orgasmo52.x
Have another 15-minute cup of coffee. It’s a great time to consider a modest donation to the Nerd Vittles project. You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN!
Now run passwd-master1. Set your FreePBX passwords to something very secure but different from your Linux root password.
Next, type status2 and press Enter. Write down the IP address of your new server.
If you’re using IPkall, now’s the time to log in to your hardware-based firewall/router and map UDP port 45693 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don’t worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you’re using SIPgate instead of IPkall.
For good measure, we recommend you reboot your server at this point. The command to type is simple: reboot4
Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone, and you’ll find lots of recommendations on Nerd Vittles. For today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 82812661 as the password for extension 701 and the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.
Don’t Forget! After you change your extension passwords later in this tutorial, you will need to update the password entry in X-Lite, or you will no longer be able to place calls! In fact, you will get locked out of your server for 90 minutes after three failed password attempts. So put this on a sticky note so you don’t forget, or you’ll regret it in about 15 minutes.
Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. And there’s really no reason you can’t use both if they’re available in your location. Do NOT use special characters in your provider passwords, or nothing will work! Continue reading whichever section below applies to you.
Configuring SIPgate. If you live in the U.S. and have a cellphone, we’d recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You’ll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don’t worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn’t matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you’ll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You’ll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.
Configuring IPkall. If you’ve opted to use IPkall, here’s the drill. First, you’ll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we’ll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:
We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don’t have an available number, your request apparently goes in the bit bucket. You’ll know because IPkall typically turns these requests around in a few minutes. Don’t worry about the mothership entry. We’ll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it’s dynamic, then you’ll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you’ve got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.
When you receive your new phone number, you’ll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:
It’s worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.
We’re going to be making some entries in FreePBX which is the web-GUI that manages PBX in a Flash. For now, we simply need to enter your new IPkall phone number so that incoming calls to your IPkall number will actually ring on your softphone. Later, we’ll make some further adjustments once we get Google Voice humming along.
Using a web browser from your desktop, log in to FreePBX 2.6 at the following link substituting your server’s private IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master).
When FreePBX loads, choose Setup, Trunks, ipkall (iax). In the USER Context field, enter your 10-digit IPkall phone number. Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.
TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it’s a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.
Now let’s test your new phone number. Call your IPkall number from a cellphone or some other phone. Your softphone should ring. Answer the call, and be sure you have voice in both directions! Do not proceed without success here, or the rest of the adventure is a waste of your time.
Configuring Google Voice. Google Voice still is by invitation only so the first thing you’ll need is an invite. If you’re in a hurry, then stroll over to eBay where you’ll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you’ve chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.
Google then will place a call to your number and ask you to enter a confirmation code that’s been provided. When your cellphone (SIPgate) or softphone (IPkall) rings, answer it and punch in the number. Wait for confirmation. Then hang up.
As we mentioned earlier, there’s no reason you can’t set up both SIPgate and IPkall forwarding numbers in Google Voice. Just repeat the drill with the other provider’s number if you wish to activate both numbers for use with Google Voice. They’re not both going to ring simultaneously as you will see in a minute.
While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:
- Call Screening – OFF
- Call Presentation – OFF
- Caller ID (In) – Display Caller’s Number
- Caller ID (Out) – Don’t Change Anything
- Do Not Disturb – OFF
Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.
Finally, place a test call to your new Google Voice number and be sure your cellphone or softphone rings. Don’t move forward until you’ve been able to successfully place a call to your phone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.
Adding Your Credentials to PBX in a Flash. We’re ready to insert your Google Voice credentials and SIPgate/IPkall number into PBX in a Flash. You’ll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate or IPkall RingBack DID (beginning with a 1). Don’t get the 10-digit GV number mixed up with the 11-digit SIPgate/IPkall RingBack DID, or nothing will work. 🙂
Log back into your server as root and issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!
Configuring FreePBX. Now shift back to your Desktop and, using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master). Depending upon which intermediate provider you’re using, do the following:
SIPgate Setup. When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:
7004484f0:B8TTW3@sipgate.com/4155201234
Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.
SIPgate and IPkall Setup. While still in FreePBX with your browser, click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate or IPkall Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.
Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.
Securing FreePBX. You’re almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.
Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.
Don’t forget to adjust your X-Lite password to match the password entry you made for extension 701!
Orgasmatron Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let’s place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!
Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.
Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then, in the subsequent articles below, we’ll cover in detail how to customize every application that’s been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we’d recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.
The VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.
Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. Using your new softphone, begin your adventure by dialing these extensions:
- D-E-M-O – Nerd Vittles Orgasmatron Demo (running on your PBX)
- 1234*1061 – Nerd Vittles Demo via ISN FreeNum connection to NV
- 17476009082*1089 – Nerd Vittles Demo via ISN to Google/Gizmo5
- Z-I-P – Enter a five digit zip code for any U.S. weather report
- 6-1-1 – Enter a 3-character airport code for any U.S. weather report
- 5-1-1 – Get the latest news and sports headlines from Yahoo News
- T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
- F-A-X – Send a fax to an email address of your choice
- 4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
- M-A-I-L – Record a message and deliver it to any email address
- C-O-N-F – Set up a MeetMe Conference on the fly
- 1-2-3 – Schedule regular/recurring reminder (PW: 12345678)
- 2-2-2 – ODBC/Timeclock Lookup Demo (Empl No: 12345)
- 2-2-3 – ODBC/AsteriDex Lookup Demo (Code: AME)
- Dial *68 – Schedule a hotel-style wakeup call from any extension
- 1061*1061 – PBX in a Flash Support Conference Bridge
- 882*1061 – VoIP Users Conference every Friday at Noon (EST)
Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you’ll want to activate CallerID Lookups in FreePBX. Choose Setup, CID Superfecta, Default and enter the maint password you created with passwd-master. Then choose Tools, Module Administration, CallerID Lookup, Enable, Process and Save the Settings. Then edit each of the Inbound Routes and choose CallerID Superfecta as the CID Lookup Source. Save your changes. Finally, choose Setup, CallerID Lookup Sources, CallerID Superfecta and be sure your maint password created with passwd-master is correct here, too. If not, update it. For additional tips, visit the forums.
Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you’ve heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It’s also perfect for off-site backups!
Also check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches for Orgasmatron 5.1 and 5.2. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.
Coming Attractions. In our next episode, we’ll walk you through the process of adding a second, third, fourth, and fifth Google Voice line to your server so that you’ll never run out of free calling on your server. Enjoy!
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
- passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. [↩]
- status is the PIAF utility program that displays the current status of most major applications running on your server. [↩]
- Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device "inside" your firewall for further processing. [↩]
- reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. [↩]
Introducing PogoPlug: Cloud Computing for $100 per Terabyte
Ever wished you could build and manage your own Cloud Computing Center with minimal cost and no recurring charges… ever? Well, today’s your lucky day.
It takes a lot to get us excited about a new product offering. But this one is a real winner! For under $130, Cloud Engines provides you your very own PogoPlug 2.0 device that connects to your router and shares up to four USB drives over the Internet. At today’s prices and ignoring sales tax, that means you can put eight terabytes of Cloud Storage on line for a one-time cost of about $100/terabyte. To give you a point of reference, Google will rent you the same space for $256/terabyte… per year. And Google is one of the least expensive Cloud Computing resources out there. Here’s the math for naysayers:
4 – WalMart1 2TB WD MyBook Drives @ $169 each = $676
1 – PogoPlug 2.0 Device @ $129 each = $129
ONE-TIME, NON-RECURRING COST: $805/8TB or $100/TB
For those that don’t need 8 terabytes, the 2 terabyte setup including the drive and PogoPlug device is still just over half the one-year rental rate of equivalent storage from Google. And, just to be clear, this isn’t merely a storage device (like Amazon S3) requiring downloads before the files can actually be used. PogoPlug’s software makes these USB drives an integral part of your Desktop just like any other attached storage devices. Think WebDAV! So it makes a perfect home for your music, movie, and photo collections. There also are loads of Open Source applications for PogoPlug for those that like to tinker. And you can use PogoPlug to keep synchronized backups of your important files.
Other Options. Be aware that for about $50 less, you can purchase the Seagate FreeAgent DockStar Network Adapter which includes a single year of PogoPlug Internet support. After that, it’s $30 annually. Translation: By the end of the second year, you’re better off with the PogoPlug. So the choice is a No-Brainer in our book. But, the fact that Seagate is also standing behind the PogoPlug design should make everyone sleep more soundly.
Deployment. After a one-minute, one-time setup over the Internet, you can securely access all of your USB drive resources via PogoPlug using either a web browser or one of several free desktop applications that are available for Windows, Mac OS X, Linux as well as Android phones, iPhones, and (earlier today) Blackberrys. And you get free support and a terrific forum. The device works flawlessly behind either a DSL or cable modem AND a NAT-based router so there are no firewall issues to address. Just enter the serial number on the bottom of your device when you access the PogoPlug web site, and configuration is automatic.
Uploading Files. One of PogoPlug’s slickest features is its automatic cataloging of files which are uploaded. Once uploaded, you can view your Music, Movies, and Pictures by simply clicking on one of the buttons. Photos are cataloged into directories by the month in which the photos were taken. Music is indexed by artist, album, and genre. In addition, music by artist, album and genre as well as photo albums can be shared by entering email addresses for those that can access the materials, by enabling public viewing (assuming you have legal rights to do so), or by sharing items using your Twitter, Facebook, and MySpace credentials. We’ve shared a photo album just to give you an idea of how this works. The security and logistical nuts and bolts all are managed by Cloud Engines’ servers. You can review and modify the materials you’re sharing by clicking on the Files I Share link in your browser. Finally you can automatically alert those with share privileges when folder content is updated. Very slick!
Give PogoPlug a try. By clicking on one of our links, you also help support the Nerd Vittles project. We think you’ll be as thrilled as we are with this terrific new creation. Enjoy!
Need help with Asterisk®? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
- The in-store pricing at WalMart is actually cheaper than on line for these particular drives. [↩]
Tweaking Asterisk for Free Google Voice Calling
Now that the Asterisk® and Google Voice marriage is finally underway, we wanted to step back today and revise the original methodology a bit to take advantage of some of the terrific comments which were offered in response to our last article. First, the good news. U.S. calls through Google Voice using Asterisk work! They sound great, and they're free. The not so good news was that the MeetMe conferencing trick to join your outbound call with the Google Voice click-to-dial return call from your destination worked great so long as a real person answered the phone. But, if an answering machine picked up or no one answered the call at all, there were problems because these calls already had been transferred to the MeetMe conference and there was no simple way to disconnect them. And the need for two DIDs to support a single Google Voice interface just seemed a bit wasteful.
9/1/2010 Update: A good bit has changed with Google Voice since this article was first published. For the definitive guide and installation procedure, we highly recommend The Incredible PBX and accompanying article which can be found at this link. Google Voice (and much more) already is included in our new PBX which is literally Plug-and-Play. If you prefer to roll your own, be sure to also have a look at this excellent update on the Michigan Telephone Blog.
Today we want to try to eliminate these two quirks while stiill providing a seamless interface between Google Voice and Asterisk. We also appreciate that thousands of you already have implemented the previous approach. So we want your transition to the new way of doing things to be as painless as possible. On the other hand, for frequent readers, we hope you'll bear with us as we repeat some of what already has been covered in previous articles so new visitors don't have to jump around between articles to get the complete picture of what we're trying to accomplish.
The objective remains the same. We want a methodology that lets us make outbound calls from any Asterisk phone using the Google Voice service to take advantage of free calling in the United States and Canada. And we want calls to our Google Voice number delivered to our Asterisk system for transparent call processing. Yes, SIP is still on our wish list for both outbound and inbound calls with Google Voice, but we'll make do with PSTN calls particularly while Google is footing the bill for all of the calls.
Update: There's now a turnkey Asterisk solution that implements Google Voice calling without getting your hands dirty. Check out our new Orgasmatron V.
Tweaked Design. Here's the new design. You obviously still need a free Google Voice account. If you don't have one, you can request an invite here. At last report, it's only taking a few days from application to invite which is really great news. Don't use a space in your Google Voice password! Once you have a Google Voice account and phone number (Google has reserved a million of them so... not to worry!), then you'll need a DID that provides unlimited, free incoming calls. Once you get your DID set up on your Asterisk system, we'll set up a forwarding phone number for this DID in your Google Voice account so that Google Voice calls can be connected to your Asterisk server.
For outbound calls, we'll combine a little dialplan voodoo with pygooglevoice to instruct Asterisk to place a click-to-dial call using your Google Voice forwarding number. Then we'll stuff in the destination U.S. phone number. When you dial GV-678-1234567 from any of your Asterisk phones, Asterisk will park your initial call in a reserved parking lot slot and then join the called party to the originally parked call. The entire procedure is virtually transparent both to the caller and the callee. And, unlike the MeetMe conference, the parking lot fades out of the picture as soon as the call is connected. Thus, if either party hangs up, the active channel for the call is terminated on your Asterisk server.
For inbound calls from your Google Voice number, we'll tweak the dialplan so that it can distinguish between a RingBack call that Google Voice initiated and a true inbound call. We'll peel off the real inbound calls and route them to a separate Inbound Route in FreePBX for processing in any way you desire.
Finally, for those that implemented the methodology in our previous article, we'll walk you through the steps to revise your existing setup to take advantage of these new tweaks. You can skip over the initial installation process if you already have gone through the Google Voice setup from our earlier article. Just skip down to Tweaking Previous Setups.
Special Thanks. At the outset, we again want to express our sincere appreciation to Jacob Feisley and Paul Marks for their pioneering work on a Python interface to Google Voice. We also stumbled upon another Python development project, Google Voice for Python. While we originally had planned to rely upon Jacob and Paul's script, we ultimately decided to implement pygooglevoice because of the additional flexibility it provided for down the road. With pygooglevoice, you not only can make Google Voice calls, but you also can send SMS messages with no muss or fuss. Jacob Feisley has now joined that project as well. So, our special tip of the hat goes to the entire Google Voice for Python development team. It's a terrific product as you will see.
Prerequisites. Today's setup requires a CentOS-based Asterisk aggregation with a current version of FreePBX. Be aware that today's solution requires Python 2.4 or higher and reportedly will not work with Python 2.3 found in some Linux distributions. We've tested everything with PBX in a Flash and, on that platform, you're good to go. The install script should work equally well with the other CentOS-based Asterisk aggregations, but we haven't tested them. Be our guest, and let us know if you encounter any problems. Finally, a word of caution. We don't ordinarily distribute solutions using development tools we don't use. Our knowledge of Python wouldn't fill a thimble. We've made an exception today because of the extraordinary interest in Google Voice by the Asterisk community. But, if something comes unglued, we can't fix it. So have a backup plan in place just in case. 🙂
Today's Drill. To get everything working today, there are six steps: (1) obtaining and configuring a DID to manage calls between Google Voice and Asterisk, (2) configuring a Google Voice forwarding number for this DID to manage your outbound and inbound calls, (3) configuring FreePBX to route all outbound calls with a GV prefix to your special Google Voice dialplan context, (4) configuring an inbound route to manage incoming calls from your Google Voice number, (5) setting up a series of Parked Call extensions, one of which will be used to manage your outbound Google Voice calls, and (6) running our install script which adds the dialplan code for Google Voice calling with your credentials and puts the Python application into place on your server. It sounds more complicated than it is. So hang on to your hat. Here we go!
Dedicated DID. Before you can use Google Voice with Asterisk, you'll need a DID that can be dedicated to your Google Voice interface to Asterisk. We'd recommend a free IPkall or SIPgate DID. To get started, use one of the links above to obtain and configure the DID. Temporarily point the DID to an extension on your Asterisk system that can be used to verify your requests for the number. Since all of these calls are free, the area code of the DID really doesn't matter because you're never going to publish the fact that it exists.
The easiest method for setting up the DID is to first create a SIP URI for the DID on your Asterisk system. Next route the SIP URI to an Inbound Route in FreePBX where you can manage the destination for calls to that DID. Initially, you want the destination to be an extension on your Asterisk system that you can answer to verify both the DID setup and the GV setup below. Finally, point the DID you obtained to the SIP URI defined above.
HINT: The entry in extensions_override_freepbx.conf would look something like this for a SIP URI called ipkall-1:
exten => ipkall-1,1,Goto(from-trunk,${DID},1)
Then you would create an inbound route named ipkall-1 using FreePBX and designate some existing extension on your server as the destination for these inbound calls.
When you set up the SIP forwarding for the DID at ipkall.com, you'd specify the SIP URI as:
ipkall-1@ipaddress_of_your-Asterisk_server
We've previously covered in detail how to do this so read the article if you need a refresher course. To reiterate, the area code of this DID really doesn't matter because you're never going to give out the number. So use one of the free sources and save yourself some money. The real trick is you want to use a DID with unlimited, free inbound calls. Both IPkall and SIPgate provide that functionality at no cost.
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of your DID. Be sure the DID is pointed to an extension on your PBX that you can answer since you have to go through Google's confirmation drill to successfully register the number. After the DID is confirmed, be sure there's a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to your Asterisk server.
While you're still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller's Number. Be aware that IPkall DIDs only forward your IPkall number as the CallerID number while SIPgate DIDs reportedly forward the actual number of the person calling you. If this matters to you, then you may prefer the SIPgate DID option. Finally, uncheck Do Not Disturb. Now click the Save Changes button.
Integrating Google Voice into Asterisk with FreePBX. Open FreePBX with a web browser and choose Setup, Trunks, Add Custom Trunk. Insert your GV number in the Outbound CallerID field and add the following Custom Dial String on the form and Submit Changes and reload the dialplan:
local/$OUTNUM$@custom-gv
Next, choose Setup, Outbound Routes, Add Route and fill in the following entries on the form:
Route Name: GoogleVoice
Dial Pattern: 48|NXXNXXXXXX
Trunk Seq: local/$OUTNUM$@custom-gv
Inbound Routes. Next, we need two Inbound Routes to get everything working. In setting up your DID with IPkall or SIPgate, you already should have created one inbound route for that provider. It already should be routing calls to an extension on your PBX. Now we need to create a Custom Destination for this inbound route and then reroute these calls there. In that way, your RingBack calls will be routed to some special dialplan code that drops these calls into a custom parking lot where the RingBack call is married up to the extension from which you placed the original call. Then we need to create another inbound route to manage normal incoming calls that are forwarded to your PBX whenever someone dials your Google Voice number.
To begin, choose Tools, Custom Destinations, Add Custom Destination and add an entry like this and then click the Submit Changes button:
Custom Destination: custom-park,s,1
Description: Custom GV-Park
Next choose Setup, Inbound Route and click on the inbound route you created previously for IPkall or SIPgate. Change the destination for these calls to Custom Destination: Custom GV-Park.
Now click on Add Incoming Route and create a new route for your incoming Google Voice calls. Give it any description you like but, for the DID number, it must be gv-incoming. You can leave most of the other defaults. Just be sure you set a destination for your incoming calls from Google Voice. It could be an extension, ring group, IVR, or whatever best meets your needs. The important entry here is gv-incoming for the DID number. Click the Submit button to save your entries. Ignore the warning that you've entered an oddball DID. We know what we're doing. 🙂
Setting Up the Parking Lot. While still in FreePBX, we need to create or adjust your existing settings in Setup, Parking Lot. The parking lot is used by FreePBX to simulate old key telephones where you could place a call on hold and then someone else in the office could pick up the call by clicking on the blinking key on their phone. The Asterisk equivalent is to press the flash hook and dial your Parking Lot Extension which then places the call in a Parking Lot space and tells you what the space number is. Someone else then can dial the number of that space to pick up the call. Our little trick today works like this. When you place an outbound call through Google Voice, your extension will be dumped into a reserved parking lot space. When Google Voice initiates the RingBack call before connecting the destination number you've dialed, that call will be sent to the same reserved parking lot space. The two calls then are joined, and you'll hear the parking lot number followed by ring tones as your call is connected by GV to its final destination. Our special thanks to Richard Bateman for his comment on the previous article and this terrific tip! He wins an Atomic Flash installer from Nerd Vittles. In addition, A. Godong wins an Atomic Flash installer for his tip on consolidating two DIDs into a single DID to manage both inbound and outbound GV calls. Just send us your addresses.
Now, where were we? Most FreePBX systems have a default setup for the Parking Lot. What we need to do is be sure you have reserved one more space in the parking lot than you actually need for day to day operation of your PBX. We'll use the last parking lot space number to manage outbound calling through Google Voice. Our entries look like the following:
Enable Parking Lot Feature: checked
Parking Lot Extension: 70
Number of Slots: 5
Parking Timeout: 30 seconds
Parking Lot Context: parkedcallsDestination for Orphaned Calls: Terminate Call: Hangup
If you use our setup above, the Magic Number is 75 which is the fifth slot in the Parking Lot. If you use a different Parking Lot extension or number of slots, here's how to calculate the Magic Number. Start counting the slots beginning with one more than the Parking Lot Extension. When you get to the last slot in the number of slots you've specified, that's your Parking Lot Magic Number. Write it down. You'll need it in a second when you run our GV installation script.
Save your entries and reload the Asterisk dialplan when prompted.
Integrating pygooglevoice. Now we're ready to complete the setup by running our revised script which loads pygooglevoice and sets up your dialplan in extensions_custom.conf. You'll need 5 pieces of information to run the script so write them down before you begin:
1. Your 10-digit Google Voice phone number
2. Your Google Voice email address
3. Your Google Voice password (no spaces!)
4. Your 11-digit RingBack DID (16781234567)
5. Your Parking Lot Magic Number
A word of caution: If you used a gMail address to set up your Google Voice account, it's possible to have different gMail and Google Voice passwords. For this to work, you'll need to enter your gMail password, not your Google Voice password (assuming they're different).
Now log into your Asterisk server as root and issue the following commands:
cd /root
wget http://bestof.nerdvittles.com/applications/gv/install-gv-new
chmod +x install-gv-new
./install-gv-new
Google Voice Speed Dials. For frequently called numbers, you can add speed dials by inserting entries in the [from-internal-custom] context of extensions_custom.conf that look like the example below where 333 is the speed dial number and 6781234567 is the area code and number to call. Be sure to reload your Asterisk dialplan to activate them.
exten => 333,1,Dial(local/6781234567@custom-gv,300)
Congratulations! You now have what we hope will be flawless and free U.S. calling on your Asterisk system using Google Voice. No gimmicks, no strings, no cost. Enjoy!
Finally, one additional word of caution. Both Google Voice and this call design are set up for a single call at a time. There are no safeguards to prevent multiple calls, but that may violate the Google Voice terms of service.
Asterisk 1.6 Solution. Several readers now have documented the procedure for implementing the Asterisk 1.6 bridge technology to make outbound Google Voice calls. You can read all about it here.
Tweaking Previous Setups. If you installed pygooglevoice using our previous tutorial, here's what you need to do. First, log into your Asterisk server as root and issue the following commands:
cd /etc/asterisk
nano -w extensions_custom.conf
Scroll to the bottom of the file by pressing Ctrl-W then Ctrl-V. Move up the file using up arrow until you reach [custom-gv]. Press Ctrl-K repeatedly to delete all of the lines in the [custom-gv] context. If you get to another line that starts with a label in brackets like [this], STOP deleting. Once you've deleted all of the lines in the [custom-gv] context, save the file: Ctrl-X, Y, and press Enter.
Now continue reading this article by jumping up to the Google Voice Setup topic. The Custom Trunk entry and the GoogleVoice outbound route will already be in your FreePBX system so there's no need to repeat those two steps. You will need to perform the remaining FreePBX steps beginning at the Inbound Routes topic and continuing on with Setting Up the Parking Lot. Finally, when you run the new installation script, it will detect that pygooglevoice is already on your system and will skip that step but will install the new custom contexts in extensions_custom.conf using your new settings. Enjoy!
Thought for the Day. Which is more arbitrary: (1) Apple snubs Google Voice or (2) Google Voice snubs SIP? Pays to look in the mirror occasionally.
Best Read of the Week. Memo to Steve Jobs and Apple: Stop Being A Jerk!
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...