Category: Audio

Siriously: It’s Wolfram Alpha for Asterisk

Ever wished your Asterisk® server could harness the power of a 10,000 CPU Supercomputer to answer virtually any question you can dream up about the world we live in? Well, so long as it's for non-commercial use, today's your lucky day. Apple demonstrated with Siri™ just how amazing this technology can be by coupling Wolfram Alpha® to a speech-to-text engine on the iPhone 4S. And, thanks to Google's new speech transcription engine and Wolfram Alpha's API, you can do much the same thing with any Asterisk server. Today, we'll show you how.1

We had such a good name for this project, Iris, which is Siri spelled backwards. You know the backwards sister and all of that. Unfortunately, the new (similar) product for Android phones was named Iris two months ago. And we didn't want to be like Larry on Newhart with two brothers named Darryl. So... we give you 4747. You can figure it out from there.

When people ask what exactly Wolfram Alpha is, our favorite answer was provided by Ed Borasky.

It's an almanac driven by a supercomputer.

That's an understatement. It's a bit like calling Google Search a topic index. Unlike Google which provides links to web sites that can provide answers to queries, Wolfram Alpha provides specific and detailed answers to almost any question. Here are a few examples (with descriptions of the functionality) to help you wrap your head around the breadth of information. For a complete list of what's available, visit Wolfram Alpha's Examples by Topic. Type a sample query here. Or call our demo line2 (1-904-339-8254 or iNum: 883510009043155) and say:

Weather in Charleston South Carolina
Weather forecast for Washington D.C.
Next solar eclipse
Otis Redding
Define politician
Who won the 1969 Superbowl? (Broadway Joe)
What planes are overhead? (flying over your server's location)
Ham and cheese sandwich (nutritional information)
Holidays 2012 (summary of all holidays for 2012 with dates and DOW)
Medical University of South Carolina (history of MUSC)
Star Trek (show history, air dates, number of episodes, and more)
Apollo 11 (everything you ever wanted to know)
Cheapest Toaster (brand and price)
Battle of Gettysburg (sad day :-) )
Daylight Savings Time 2012 (date ranges and how to set your clocks)
Tablets by Motorola (pricing, models, and specs from Best Buy)
Doughnut (you don't wanna know)
Snickers bar (ditto)
Weather (local weather at your server's location)

Best Question of the Day Award: "How much wood could a woodchuck chuck if a woodchuck could chuck wood?" And the answer: "A woodchuck would chuck all the wood he could chuck if a woodchuck could chuck wood. According to the tongue twister, although the paper 'The Ability of Woodchucks to Chuck Cellulose Fibers' by P.A. Paskevich and T.B. Shea in Annals of Improbable Research vol. 1, no. 4, pp. 4-9, July/August 1995, concluded that a woodchuck can chuck 361.9237001 cubic centimeters of wood per day."

Implementation Overview. Today what we're going to demonstrate is how to configure your Asterisk server so that you can pick up any phone on your system, dial 4-7-4-7, speak a question, and we'll show you how to send it to Google to convert your spoken words into text. Then we'll pass that text translation to Wolfram Alpha which will provide a plain text answer to your question. Finally, we'll take that plain text and use Flite or Cepstral to deliver the results to you.

For openers, you'll need a free Wolfram Alpha account. We'll be using PBX in a Flash 2.0.6.2.1™ to demonstrate the setup because its reliance on CentOS 6.2 provides the most complete collection of Linux utilities available. And, of course, you get unlimited, free calling within the U.S. and Canada with Google Voice as part of any PBX in a Flash install. It's certainly possible to do what we're demonstrating on other Asterisk server platforms once you get all of the dependencies resolved. But we'll leave that for the pioneers.

Using PIAF2™, you'll need to download a new AGI script to take advantage of Google's speech transcription engine. No registration is (yet) required. Then we'll provide a simple piece of dialplan code to handle the phone conversation. Finally, we'll provide a couple of AGI scripts to tame the Wolfram Alpha interface for you. Plug in your Wolfram Alpha APP-ID, and you'll be off to the races. It's about a 15-minute project using an existing PIAF2 server. So let's get started.

Legal Disclaimer. What we're demonstrating today is how to use two publicly accessible web resources to harness the power of a supercomputer to respond to your queries using a phone connected to an Asterisk server. We're assuming that both Google and Wolfram Alpha have their legal bases covered and have a right to provide the public services they are offering. We are not vouching for them or the services they are offering in any way. By using our scripts, YOU AGREE TO ASSUME ALL RISKS, LEGAL AND OTHERWISE, ASSOCIATED WITH USE OF THESE FREELY ACCESSIBLE WEB TOOLS. NO WARRANTY EXPRESS OR IMPLIED IS BEING PROVIDED BY US INCLUDING ANY IMPLIED WARRANTY OF FITNESS FOR USE OR MERCHANTABILITY. You, of course, have an absolute right not to use our code if you have reservations of any kind or are unwilling to assume all risks associated with such use. Sorry for legalese, but it's the time in which we live I'm afraid. Plain English: "Don't Shoot the Messenger!"

Getting a Wolfram Alpha Account. As you can imagine, there have to be some rules when you're using someone else's supercomputer for free. So here's the deal. It's free for non-commercial, personal use once you sign up for an account. But you're limited to 2,000 queries a month which works out to almost 70 queries a day. Every query requires your personal application ID, and that's how Wolfram Alpha keeps track of your queries. Considering the price, we think you'll find the query limitation pretty generous compared to other web resources.

To get started, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That's all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.

One-Click Installer. If you don't care about how things work, you can skip all of the steps below and use the new one-click installer. Or you can keep reading to see what's going on. Here are the steps to use the one-click installer. Log into your server as root and issue the following commands:

cd /root
wget http://nerd.bz/xhUpJr
chmod +x wolframalpha-oneclick.sh
./wolframalpha-oneclick.sh

You now can skip the next four sections and dial 4-7-4-7 to try things out.

Installing the Google Transcription AGI Script. Log into your PIAF2 server as root and issue the following commands to download and install Lefteris Zaferis' AGI script from GitHub. It's a terrific piece of code!

cd /root
wget --no-check-certificate http://nerd.bz/w8HCDF
tar zxvf asterisk-speech*
cd asterisk-speech-recog-0.4
cp speech-recog.agi /var/lib/asterisk/agi-bin/.

If you prefer living on the Bleeding Edge, you can download Lefteris' very latest (untested by us!) release3:

cd /root
wget --no-check-certificate http://nerd.bz/zA4fCB
tar zxvf asterisk-speech*
cd asterisk-speech-recog-0.5
cp speech-recog.agi /var/lib/asterisk/agi-bin/.

Installing the Wolfram Alpha Scripts. Now log into your PIAF2 server as root using SSH and issue the following commands to install the Wolfram Alpha transportation layer:

cd /
wget http://nerd.bz/A7umMK
tar zxvf 4747.tgz
cd /tmp
cat 4747.txt

Adding the Asterisk Dialplan Module. What is displayed on your screen at the end of the steps above will be the dialplan code that needs to be added to extensions_custom.conf in the /etc/asterisk directory. Just cut-and-paste the code and drop it into the [from-internal-custom] context. If you use nano, be sure to open the file with nano -w extensions_custom.conf to avoid problems with long lines being truncated. You'll notice that there are commented lines 3, 6, 16, and 17 to support Cepstral. If you use this commercial TTS app which now can be installed in PIAF2 with install-cepstral, then you can comment out the Flite entries and uncomment the Swift (Cepstral) entries in the dialplan code. Here's the SED alternative rather than manually updating the file with cut-and-paste:

cd /etc/asterisk
cp /tmp/4747.txt .
sed -i '/\[from-internal-custom\]/r 4747.txt' extensions_custom.conf
asterisk -rx "dialplan reload"

If you manually edit, don't forget: asterisk -rx "dialplan reload".

Adding Wolfram Alpha APP-ID. The final configuration step is adding your Wolfram Alpha APP-ID credentials. Issue the following commands to access the AGI script:

cd /var/lib/asterisk/agi-bin
nano -w 4747

When the file opens, replace yourID between the quotes with the APP-ID that was provided to you on the Wolfram Alpha web site. Then save the file: Ctrl-X, Y, then Enter. You're done!

Tweaking the Abbreviations List. Translating abbreviations into speech is a tricky business, and Flite and Cepstral do a pretty lousy job on some of them. We've started the beginnings of an abbreviation list which you will find in the function section of 4747.php which is stored in /var/lib/asterisk/agi-bin. It's easy to add additional entries. Just clone one of the entries that's already there. For example, here's the line that translates Jr. into Junior. HINT: Be careful to surround most unpunctuated abbreviations with spaces, or you may get unexpected results when a word actually begins or ends with the same letters.

$response = str_replace("Jr.","junior",$response);

Taking Wolfram Alpha for a Spin. Some sample commands have been documented above to get you started. Just pick up a phone on your PIAF2 server and dial 4747. When prompted, say one of the commands and press the pound key. Your command will be sent to Google for translation, and then the text result will be played back using Flite or Cepstral. If it says what you meant to say, press 1 to launch the Wolfram Alpha connection and get the answer to your question. If not, press * and try again.

You also can watch the progress of your calls on the Asterisk CLI. We've found the Google speech-to-text transcription to be extremely accurate in quiet rooms. One of the variables returned in the [4747@from-internal:5] entry on the Asterisk CLI includes a transcription accuracy measurement which is shown as a decimal number less than 1. This gives you an idea of how well Google is understanding your accent. If the number consistently falls below .9, you may want to move out of the Deep South for a bit. :wink:

Originally published: Monday, January 16, 2012




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


Some Recent Nerd Vittles Articles of Interest...

  1. We want to extend a special welcome to our Hack A Day and Reddit visitors. We have new tips and tricks on VoIP technology every week. And almost half of our traffic is from returning visitors. We hope you'll join the club. Thanks for visiting. []
  2. Because of a few "special people" we've had to limit calls to one per person. You still can beat the system by calling back from a different phone. :wink: For those that are curious, this demo line is supported by Google Voice so you can check out the call quality for yourself. We alternate hosting the trunk on either an Aspire Revo or one of 10 PBX in a Flash servers running as virtual machines under Proxmox on a $500 Dell PowerEdge T310 server behind a secure, hardware-based firewall with no Internet port exposure and no ports forwarded from the firewall to the server. Dell servers go on sale about once every couple of weeks. []
  3. Version 0.5 also includes some sample Wolfram Alpha perl code that is certainly worth a look. []

11/11/11: To Celebrate Nerd New Year’s, Please Welcome…

Nerd Vittles Daily Dump

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.


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


Some Recent Nerd Vittles Articles of Interest…

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

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

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

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

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

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

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

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

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

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

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

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

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

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

NOTE: If xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for CentOS 5.x systems:

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

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

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

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

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

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

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

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

./skype &
cd /siptosis
./SipToSis_linux

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

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

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

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

Now reboot your server and make sure everything is working.

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

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

2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To find the Skype PID: pidof skype. To kill Skype: kill pid#. To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_Linux

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

Setting Up Speed Dials for Skype Friends. One of the wrinkles with Skype is that Skype uses names for its users rather than numbers. If you don't have a SIP URI-capable softphone, there's still an easy way to place calls to your Skype friends using FreePBX®. Just add a Speed Dial number to your FreePBX dialplan. Choose Extension, then select the Custom type, provide an Extension Number which is the Speed Dial number (this could actually spell your friend's name using a TouchTone phone), enter a Display Name for your friend, and add an optional SIP Alias. Then insert the following in the dial field replacing joeschmo with your friend's actual Skype name. Save your entries and reload the dialplan when prompted.
SIP/joeschmo@127.0.0.1:5070

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

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

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

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

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

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

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

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

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

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

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

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

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

asterisk -rx "dialplan reload"

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

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

Originally published: Tuesday, November 1, 2011


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.


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


Some Recent Nerd Vittles Articles of Interest...

  1. 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.


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


Some Recent Nerd Vittles Articles of Interest…

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.


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


Some Recent Nerd Vittles Articles of Interest…

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

Incredible PBX for Asterisk 1.8: Back from the Brink

Photo courtesy of<br />
Blou_Aap. Artist unknown but terrific.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.

Prerequisites. Here's what we recommend to get started properly:

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. :roll: 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. :wink: 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. :wink:

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.



Click above. Enter your name and phone number. Press Connect to begin the call.


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

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.


Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won't have to wait long for an answer to your questions.




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


whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.


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


Some Recent Nerd Vittles Articles of Interest...

  1. For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. []
  2. 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 []
  3. HINT: Version 1.7.5.5.3 ISO also works just fine. []

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