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The Most Versatile VoIP Provider: FREE PORTING

Lessons Learned: Getting Started in the Billion Dollar VoIP Business

So you’ve built a few VoIP PBXs for your neighbors and your friends’ small businesses. And now you want to make a living doing it full time. After all, it wasn’t that hard to get started since all of the VoIP software was practically free, and the hardware investment was only a few hundred bucks. But now your friends need a way to make reliable phone calls every day, and they want someone to call when the phones don’t work. Welcome to the VoIP Business! Our objective today is to paint you a picture of what actually lies ahead in the Asterisk® and FreePBX® business so that you don’t get blindsided.

Lesson #1. Asterisk is a business run by Digium to make money for the corporation. FreePBX is a business run by Schmooze Com to make money for the corporation. Both companies do this in several ways. They sell hardware. They sell commercial software. They sell hosted phone service. They sell phone trunks to make and receive phone calls. And they sell support. The lifeblood of these companies is paying customers, lots of them. There’s nothing necessarily sinister about any of this. It’s the way all corporations work.

Lesson #2. You can’t do it all. You may be a super salesman, a talented programmer, or a great customer service guy. But you’re probably not all three. And, if you have a family, the rest of them probably don’t want the phones ringing off the hook starting at dinner time until 2 a.m. every morning. There’s a reason corporations charge a pretty penny for support. Somebody has to be there during dinner time and at 2 a.m. to answer the phone calls and solve the problems.

Lesson #3. Your friends are cheap frugal. They’d prefer to pay nothing for their phone system, and they’d prefer to pay nothing when they need to call you to fix it. You’re a nice guy so you don’t want to leave your friends in the lurch when you decide to take that Christmas ski trip. What to do? Hire an outside company to provide your support. Heh! Keep reading.

Lesson #4. The stark reality at the corporate end of the VoIP business is RECURRING REVENUE. They can’t stay afloat just selling hardware and software. Once folks have bought it, the company either needs new paying customers or a way to keep existing customers paying to keep the lights on. There are three options: hosted phone service, phone trunks, and support.

If you’ve done your homework, you know that you can buy incoming phone lines for your PBXs at a monthly cost of a few bucks. Or you can stick with Ma Bell for incoming trunks and up the monthly cost by a factor of ten in exchange for reliability and support. Outgoing phone calls can be made for a penny or two a minute to all but the most exotic and remote areas of the world. Or you can use trunks provided by Ma Bell or Comcast or Time Warner for ten times the monthly cost. Then there are the so-called unlimited trunks from companies such as Digium and Schmooze Com. For $20+ to $25+ per month, you get the ability to make or receive several thousand minutes of calls each month so long as the calls arrive one at a time. If you want to make or receive multiple calls simultaneously, multiply the cost for each simultaneous call by twenty to twenty-five bucks depending upon your provider choice. All of a sudden, Ma Bell isn’t looking that expensive, is she?

Lesson #5. When you’ve grown your user base to the point that you don’t want to lose your customers, be careful in choosing a company to provide your support. If they happen to be in the same business as you (and they probably are), ask yourself this question. Would you send your girlfriend alone on a two-week cruise with any of your male buddies? Didn’t think so. Reread Lesson #1.

To be continued… Happy New Year!!

Originally published: Monday, December 29, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…

Santa’s Technology Roundup: The Best Products of 2014 with Some Surprises

Once a year we like to pause and take a look back at 10 technology products that really grabbed our attention. 2014 will be remembered as a spectacular year. So here’s what made the Nerd Vittles short list for 2014…

Smartphone of the Year: It’s a 5-Way Tie

And the winners in no particular order… Galaxy Note 4, iPhone 6+, LG G3, HTC One M8, and Moto X.1 So which should you choose if you can only have one? Visit AndroidHeadlines.com for a detailed feature comparison. You can’t go wrong with any of them. In our family, there’s one of almost all of them.

Desktop Computer of the Year: Apple’s 27‑inch iMac with Retina 5K display

If you work with a computer for a living, there is no competition. It scales to any feature set you may need. Run, don’t walk, to your nearest Apple Store and get in line. We waited two months for ours!

Portable Computer of the Year: Apple’s MacBook Air with Retina Display

Hah. Just kidding. It would have been the hands-down favorite in 2014 except for one minor detail. It hasn’t been released… yet. If you absolutely have to have a retina display-quality notebook, then you’ll have to settle for the slightly thicker Macbook Pro this Christmas. For us, we’re waiting for 2015 and what will surely be the MacBook Air with Retina Display.

Tablet of the Year: iPad Air 2

If you’re starting to think we’re charter members of the Apple FanBoy Club, then you haven’t been following Nerd Vittles for very long. We can be one of their harshest critics. But the bottom line is that Apple products are compelling because of their tight integration to Apple’s closed society. If you’re a member of that club, then you’ll want the iPad Air 2 to add to your collection. It’s a terrific tablet at a compelling price.

Multimedia Device of the Year: Roku 3

If you’re into Netflix and Amazon Prime and movies, nobody needs to tell you that the streaming device hardware market is a crowded place. The Roku 3 isn’t the cheapest device in the market, but it’s still the one we always drop into our suitcase when we hit the road. It’s simple to configure and supports WiFi almost anywhere. It just works!

VoIP Product of the Year: Vitelity’s vMobile

It’s taken a few starts and stops to get the kinks out, but Vitelity’s vMobile smartphone is a truly revolutionary offering. It provides seamless integration of the smartphone into your PBX infrastructure. The phone becomes "just another extension" on your PBX except the device is 100% mobile which means it works with WiFi or it works anywhere Sprint has a tower. For any organization with staff that travels, this is a must-have device. Anything you can do with a traditional PBX extension, you can do with your smartphone using the vMobile technology. It’s the hands-down winner as VoIP Product of the Year. Use our special signup link and help support the Nerd Vittles, PBX in a Flash, and Incredible PBX projects.

VoIP SOHO Hardware of the Year: CuBox-i

We’ve tested lots of small footprint hardware in search of the perfect VOIP platform for the home or SOHO office. The search is over. The hands-down winner is the CuBox-i. It’s tiny, powerful, quiet, and has every feature you could possibly want in a VoIP server. Read our full review here. They’re 25% at NewEgg if you hurry.

VoIP Deal of the Year: $15 Pogoplug with Incredible PBX

If there’s one thing all of us have in common, it’s a burning desire to find the best bargain on the planet. In the VoIP marketplace, look no further than here. Repurposing a PogoPlug for less than $20 (and some of them went for $5), is the perfect way to learn about VoIP without breaking the bank. Our tutorial on the VoIP Deal of the Year will tell you everything you need to know to get started.

Must-Have Product of the Year: Amazon Echo

The Amazon Echo is still an invitation-only device, but you need to get in line NOW. During the introduction, Amazon is selling them for $99. Or you can get one on eBay for about triple that amount. It’s money well spent. Think of it as a desktop version of Siri. But it’s so much more. With Amazon Prime and Prime Music accounts plus a free iHeartRadio account, you get access to a collection of over a million songs just by saying the name of the artist or song or playlist or radio station of interest. You also can upload 250 of your own songs not purchased through Amazon Music at no charge. Or, for $25 a year, you can upload up to 250,000 tracks much like iTunes Match. The sound quality of the device is nothing short of spectacular. My teenage daughter and I spent over two hours playing with it the first night it arrived. And the excitement hasn’t waned. It’s the go-to device for all of our visitors to explore new and old music. And, yes, Amazon Echo knows the weather, the time, and just about anything else you care to ask about. You’ll have it in your living room in no time. Not only will it speak the results while playing your favorite song, it’ll send the results and to-do list to your smartphone.

2014: Cloud Computing Reinvented

Over the past few years, we’ve seen a gradual migration of server platforms to the cloud thanks in large part to ever falling prices on the Amazon EC2 platform. But 2014 saw some new cloud strategies. First came the pay-once-use-it-forever platform of CloudAtCost.com. Wait for the next sale and save half on almost any of their server platforms. If you follow us on Twitter, we’ll let you know when it happens. We’ve had several servers for almost a year with no hiccups. In fact, we now keep backup images of the Nerd Vittles, PBX in a Flash, and Incredible PBX web sites running 24/7 on these Canadian servers. Check out the performance for yourself.

Then there was Digital Ocean with its pay-by-the-hour pricing coupled with the ability to create virtual machines for almost any platform in under a minute. It truly is a developer’s dream come true. Frankly, it’s our platform of choice for development of all the great software you read about here. Use our signup link and get a $10 credit to try things out. The beauty of the technology is you can create a server with 512MB of RAM and a 20GB drive, work for a half a day, take a snapshot of your project, and then delete the server until you feel like working again. Total cost for use of the platform and storage of your snapshot: about 2¢.

With any great new technology, of course, competition is not far behind. Meet Vultr, the Digital Ocean knock-off promising more memory, more server locations, and more features for less money. Is Vultr really better? We’ll let you know after we’ve had more time to play. Our first look uncovered a few wrinkles. First, you had to request enabling of port 25 for outbound SMTP mail support. Not a big deal if it were documented that you had to request it, but it isn’t mentioned anywhere on the site. Second, virtual machines take a bit longer to create and much longer to become fully functional on Vultr. We got spoiled by the one-minute spin up at Digital Ocean. But, the good news is a penny-an-hour server gets you a gig of RAM, 20 gigs of storage, and 2 terabytes of data transfer a month for $7. And it is fast! So stay tuned for a full review and…

Merry Christmas!

Originally published: Monday, December 22, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…

  1. Some of our purchase links refer users to Amazon and other sites when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from merchants to help cover the costs of our blog. We never recommend particular products solely to generate commissions. However, when pricing is comparable or availability is favorable, we support Amazon and other merchants because they support us. []

Incredible PBX on Steroids: The Asterisk-GUI Project Rolls On (Chapter 3)

We’re big fans of the new hybrid vehicles especially the Cadillac ELR. It combines an electric motor with a gas generator to give you the best of both worlds. For in-town driving, you get terrific performance at 1.5¢/mile using pure electric power. But you’re not hamstrung from venturing out to anywhere you choose using a traditional gas engine that can be refueled quickly at any time. In a nutshell, that’s the design philosophy that’s inspired development of Incredible PBX for the Asterisk-GUI.

This is the third installment in our series. You can catch up with the Overview as well as Chapter 1 and Chapter 2 here.

With Incredible PBX for the Asterisk-GUI, you get a terrific GUI to manage Asterisk® while taking advantage of all the neat features that Incredible PBX brings to the table using traditional dialplan design. Stated another way, you’re not being forced to always use a GUI to manage your Asterisk server when command-line utilities are more functional or efficient. Previous approaches to GUI-only management of Asterisk forced you to always jump through the GUI hoops to do much of anything. Unfortunately, what you lost in that scenario was a lot of the native functionality of Asterisk.

That’s not to say there wasn’t a lot to like about our GUI heritage with Asterisk. These open source projects brought a wealth of features to the table for beginners without having to learn much about the way Asterisk actually worked. The downside was you didn’t learn much about the way Asterisk actually worked. On the one hand, it kept folks from making serious programming errors that could result in major phone bills when security issues crept into a dialplan. The drawback was you never learned why. When something came unglued and things do come unglued, you were up the proverbial creek without a paddle. In fact, many never knew they had a paddle much less what it looked like.

I wish I had a nickel for every user that’s complained over the years that Asterisk won’t start. The last time we checked Google was showing 963,000 of them. It turns out that many of these weren’t failures with Asterisk at all but rather shortcomings in the interaction of one particular graphical user interface with MySQL. If you don’t believe it, shut down MySQL on your existing Asterisk server and then try to restart Asterisk. You’ll never see this with Incredible PBX for Asterisk-GUI. Why? Because the reliability of Asterisk isn’t tied to the reliability of MySQL, Apache, Perl, Asterisk-GUI, or any other foreign application.

**** WARNING: ERROR IN CONFIGURATION ****
astrundir in '/etc/asterisk' is set to  but the directory
does not exists. Attempting to create it with: 'mkdir -p '
mkdir: missing operand
Try 'mkdir --help' for more information.
**** ERROR: COULD NOT CREATE  ****
Attempt to execute 'mkdir -p ' failed with an exit code of 1
You must create this directory and the try again.

In the new Incredible PBX design, we haven’t forgotten about security either. In this day and age, it’s the single most important feature of any PBX that is connected to the Internet. We always recommend running your server behind a hardware-based firewall with no Internet port exposure, but we appreciate that’s not always possible particularly with Cloud-based servers. Incredible PBX is delivered with the Linux IPtables firewall preconfigured. It allows access from your server’s IP address, from the IP address used to install Incredible PBX, from private IP addresses on your local area network, and from a very limited set of trusted providers so that you can connect your trunks to make and receive phone calls. The tools to add and delete whitelist entries on your firewall are also included. In addition, we’ve included the PortKnocker utility which lets remote users with the three port knock codes gain access until their IP addresses can be whitelisted by an administrator. In addition to IPtables security, there’s another layer of protection for web-based applications. Asterisk-GUI, of course, has its own security system that’s tied to the Asterisk manager.conf setup. All of the remaining web applications require Apache authentication. For Reminders and AsteriDex, you can create multiple Apache passwords for individual users or groups of users. For administrator applications, you set an admin password that’s only known by administrators.

We couldn’t help chuckling recently when one of the security sites found a vulnerability in one of the Incredible PBX applications but noted that administrator access was required to get to the application to launch the attack. That’s akin to saying your system is vulnerable if you hand out your root user credentials AND whitelist the IP addresses of the bad guys. Literally, what was documented was true, but finding security issues in software that requires root permissions for access is getting a little desperate, wouldn’t you say? Of course, one of our "competitors" wasted little time splashing it all over their web site. The vulnerability was fixed the same day it was disclosed, by the way. And it was automatically pushed out to every Incredible PBX server, all of which run industry-standard Linux operating systems. That’s the approach to system design and support our users have come to expect. Feel free to compare it to the offerings you’ll find elsewhere, commercial or otherwise. That, my friends, is what freedom of choice is all about.

The Lean, Mean (Pure) Asterisk Machine

The roadmap for the future direction of Incredible PBX continues to evolve, but let us take a moment and share our current thinking. We’ve previously mentioned that the target audience for Incredible PBX for Asterisk-GUI is hobbyists. That’s not a dirty word in our book. Nor does it mean the platform won’t be as robust and reliable as previous releases of Incredible PBX. It just has a smaller memory footprint and much faster performance. Yes, we’re using Asterisk-GUI which Digium no longer supports. But that was a marketing decision that had nothing to do with the quality of the product. It was written by some of the best brains in the Asterisk business so we’re comfortable using it as a platform. We’ve found only two bugs in beating on the software relentlessly. Outbound Caller ID on a per extension basis can be quirky. Trunk-based CallerID whether assigned at the provider end or on Incredible PBX works just fine including CallerID spoofing where permitted by the provider. The other wrinkle was Asterisk-GUI’s failure to support the [context](+) feature of Asterisk. We’ve found an easy workaround for that one as well. We just won’t use it.

The plan is to roll this out first on the CentOS 6.5 (now 6.6) platform because we view it as the most stable. Scientific Linux 6.6 works equally well. Once we get any kinks out of the code, we’ll turn our attention to Ubuntu 14 and then on to the small hardware: Raspberry Pi, BeagleBone Black, CuBox, and PogoPlug. There’s also been interest in a more internationally-friendly version, and that’s on the drawing board as well. During the rollout, we hope to complete work on moving a few MySQL-based utilities to SQLite3. We will leave MySQL in the installation mix but will turn it off to further reduce the memory overhead of the install. We also will scale back the number of simultaneous Apache sessions running since the purpose of Apache is primarily to support administrator utilities on the server. Actually, you can run Asterisk-GUI using either the native Asterisk http server or with Apache. Thanks to Bill Simon of Simon Telephonics, you’ll have both options. With simple modifications, we think we can improve the performance on memory-constrained platforms dramatically while providing a robust, high performance platform if you have the hardware to support it. We’ve also initiated discussions with Amazon to roll out a phone service using this platform for the new Amazon Echo product. So 2015 is shaping up to be another banner year in the VoIP world. We hope you’ll come join us.

This week we continue the march. We want to review some of the open source features being incorporated into Incredible PBX from the open source code base minus some of the superfluous GUI modules. For example, you can manage blacklisting of callers using nothing more than your telephone. The same is true for SMS messaging. If you can dictate an SMS message, then why type it? Bash scripts are a well-tested feature of Incredible PBX, and you’ll still find a healthy collection of them in the /root folder of your server after you complete the install. But today’s focus is what can be accomplished with Incredible PBX using nothing more than your telephone.

Blacklisting Callers with Incredible PBX

One of our old PBX favorites dating back to the Asterisk@Home days was blacklisting. This means that old girlfriends and telemarketers get routed to Zapateller with a message that your number is not in service. By default, Incredible PBX for Asterisk-GUI will automatically blacklist incoming calls without a CallerID number. You can modify this behavior if desired:

asterisk -rx "database del blacklist blocked"

If you change your mind and want to turn anonymous call blocking back on, use this command:

asterisk -rx "database put blacklist blocked 1"

We’ve retained the same feature codes to manage blacklisting of specific numbers from any phone on your system:

  • *30 – Add a number to Blacklist
  • *31 – Remove number from Blacklist
  • *32 – Blacklist last number that called

Blacklisting was all smoke and mirrors in the old GUI days. But we want you to understand how this actually works so that you can change it if you’d like. For example, instead of the Zapateller tone, you might prefer to route callers on your blacklist to Lenny (53669 on your phone) so that you waste some of the caller’s time instead of the other way around.

In the extensions_additional.conf file, find the [app-blacklist-check] context. The last four lines in that context look like this:

;exten => s,n,Goto(DLPN_DialPlanMain,53669,1)
exten  => s,n,Zapateller()
exten  => s,n,Playback(ss-noservice)
exten  => s,n,Hangup

To route blacklisted callers to Lenny, just uncomment the top line shown and add semicolons to the next two lines:

exten  => s,n,Goto(DLPN_DialPlanMain,53669,1)
;exten => s,n,Zapateller()
;exten => s,n,Playback(ss-noservice)
exten  => s,n,Hangup

Wasn’t that easy? Now just save your changes and reload your dialplan: asterisk -rx "dialplan reload"

You may prefer to manually add numbers to your blacklist. You can do this from the Linux command prompt like this. Don’t forget the 1.

asterisk -rx "database put blacklist 8005551212 1"

From the Asterisk CLI (asterisk -rvvvvvvvvvv), do it like this:

database put blacklist 8005551212 1

To display all of your blacklist entries, try this:

database show blacklist

To remove an entry from the blacklist, use this syntax:

database del blacklist 8005551212

MP3 Voicemail Messaging for Cellphone Playback

One of the most requested features on our forums has been the ability to forward voicemails in MP3 format so that they play back correctly on cellphones and desktop mail clients. As with many of the Incredible PBX features, we wouldn’t know where to start to thank all of the folks that helped make this happen. You can review the thread on the PIAF Forum for background. This is yet another great example of how the open source community should work. Thanks to everyone that participated in bringing this development to fruition. On the new Incredible PBX for Asterisk-GUI platform it’s automatic. All you have to do is assign an email address to any voice mailbox on your server in the Users setup, and incoming voicemail messages will be delivered by email in the proper format for playback. The message thread explains how for those with an interest.

Accessing Voicemail Messages with Incredible PBX

Speaking of voicemail, we’ve tried to maintain the same feature codes that many have become accustomed to over the years. Here’s a recap of the codes in case you ever forget:

  • *98 – Check Voicemail Messages from Any Phone
  • *extension – Leave a Voicemail for Dialed Extension
  • * after voicemail connect – Access Voicemail Retrieval

Migrating the Google Speech Feature Set to Incredible PBX for Asterisk-GUI

We previously mentioned that Google Voice wasn’t around when Asterisk-GUI was developed. Not to worry. We’ve added it. And that’s just the beginning. All of the Google features that have made Incredible PBX so popular will be included in the Asterisk-GUI edition. That includes text-to-speech and speech recognition thanks to Lefteris Zafiris. It also includes SMS messaging with your same Google Voice credentials. Pick up a phone and dial S-M-S to dictate and send an SMS message to any recipient in the U.S. or Canada. Pick up a phone and dial 949 to listen to a weather forecast for any major city in the world. Just say the name of the city and state or country. Pick up a phone and dial 951 to listen to the latest News Headlines. Or dial T-O-D-A-Y to listen to Today in History. Sign up for a free Wolfram Alpha key, dial 4747, and you’ve got a voice-enabled encyclopedia at your fingertips. Eat your heart out, Siri. Our extra special thanks to Google for still supporting the open source community. Did we mention… It’s all still free.

Google has changed the rules a bit on using their speech recognition engine. So you now need an API Key to use the Speech Recognition AGI script for Asterisk. Assuming you’ll be using the functionality for “personal and development use,” here’s how to obtain your API key:

1. IMPORTANT FIRST STEP: Use an existing Google/Gmail account to join the Chrome-Dev Group.

2. Using the same account, create a new Speech Recognition Project.

3. Click on your newly created project and choose APIs & auth.

4. Turn ON Speech API by clicking on its Status button in the far right margin.

5. Click on Credentials in APIs & auth and choose Create New Key -> Server key. Leave the IP address restriction blank!

6. Write down your new API key or copy it to the clipboard.

7. Once you’ve installed Incredible PBX, log into your server as root and edit speech-recog.agi in /var/lib/asterisk/agi-bin.

8. Go to line 70 of speech-recog.agi: my $key = "". Insert your API key from Step #6 above between the quotation marks and save the file: Ctrl-X, Y, then Enter.

This will activate all of the Speech Recognition applications in Incredible PBX as described above.

Activating Wolfram Alpha with Speech Recognition in Incredible PBX

If you’re not familiar with Wolfram Alpha, it’s an encyclopedia and almanac on steroids. It’s driven by a supercomputer. There’s not much it doesn’t know. We’ve written an exhaustive article on Wolfram Alpha for Asterisk so start there. With Incredible PBX, everything is preconfigured for you. All you need to do is obtain a (free) API key.

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.

Now you’ll need to insert your API key into /var/lib/asterisk/agi-bin/4747. The first line of the file looks like this: APPID="Wolfram-Alpha-API-Key-Goes-Here". Insert your API key between the quotation marks and save the file: Ctrl-X, Y, then Enter.

You’re ready to try out Wolfram Alpha by dialing 4-7-4-7 from any phone connected to your server. Here are some sample queries to get you started:

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 2015 (summary of all holidays for 2015 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 2015 Charleston South Carolina (date ranges and how to set your clocks)
iPads by Apple (pricing, models, and specs from Best Buy)
Doughnut (you don’t wanna know)
Snickers bar (ditto)
Weather (local weather at your server’s location)

Yahoo! Weather by ZIP Code Is Moving to SQLite 3

One of the more popular features of Incredible PBX has always been the ability to retrieve a Yahoo weather forecast by dialing Z-I-P and plugging in a 5-digit ZIP code for the weather report you wished to hear. This always required a MySQL zip code database to translate the zip code into a city and state for presentation to the various weather services. As part of our move to reduce the memory footprint of Incredible PBX, we are gradually removing our dependence on MySQL. In its place we’re deploying SQLite3 databases, and Weather by ZIP Code was our first successful migration. Moving the MySQL zip code database to SQLite was a snap using a terrific open source script that we highly recommend to developers. It lets you convert any MySQL database (with indexes) to SQLite 3 in seconds. Here’s the link if you ever have the need. About 5 lines of PHP code had to be modified to complete the migration from MySQL to SQLite. Not bad. For our purposes, you’ll never know the difference when you dial in for your next weather forecast.

Originally published: Monday, December 15, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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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…

Incredible PBX on Steroids: The Asterisk-GUI Dial Plan Basics (Chapter 2)

We’re making steady progress on the Incredible PBX for Asterisk-GUI project. If you didn’t read last week’s introductory article, start there. This week we’ve had to wrestle with one of the stark realities of taking someone else’s turnkey code and attempting to bolt on enhancements. As previously noted, Asterisk-GUI works all of its magic by manipulating Asterisk® config files directly with no outside storage of settings in either MySQL or the Asterisk DB. This is a good thing… at least until you try to add new features while leaving the basic Asterisk-GUI code intact. That was one of our primary objectives.

This is the second installment in our series. You can catch up with the Overview as well as Chapter 1 and Chapter 3 here.

Simplifying Credentials Management with Incredible PBX

Here’s the problem. We wanted to separate out the credentials for various providers so it would be easy for first-time users to set up a server without having to master Asterisk or the Asterisk-GUI. As we mentioned, this has been the number #1 complaint with the FreePBX® way of doing things. You almost needed to go back to college for another degree before you could make your first phone call. To get a functional VoIP server with one extension and one outbound trunk, it required creation of an extension, registration of a trunk with obscure settings that are different for almost every provider, creation of an inbound and outbound route with settings for how to actually route the calls in and out. And then there was configuration of some less-than-intuitive SIP settings. That’s before you ever start thinking about security and a firewall. Life’s too short!

But we encountered some stumbling blocks with Asterisk-GUI as well. It was rewriting our credentials_sip.conf file whenever Asterisk-GUI was actually used to add a new user (extension) or trunk. Worse yet, it was rewriting the entries incorrectly because the developers forgot about a special syntax in Asterisk that we’ll get to in a minute.

We first thought we could solve the rewriting of our config files by limiting write permissions on our new credentials files to the root user. Asterisk and Asterisk-GUI both run as the asterisk user so this would have been an easy fix. Well, no cigar. Asterisk-GUI outsmarted us by quietly aborting the update when it didn’t have ownership of our .conf files. This meant you never could use the Asterisk-GUI for much of anything, not exactly what we had in mind.

Lucky for us, one of the developers forgot about our favorite Linux utility, chattr. This lets you set the immutable bit to prevent all users (including root) from changing the contents of a file. Since Asterisk-GUI sometimes skips error checking after it’s sure it owns the .conf files, it was perfect. We could "hide" settings in our own credentials_sip.conf file without worrying that they’d be overwritten by Asterisk-GUI. The only trick is remembering to turn the immutable bit off when we want to make updates and, of course, turning it back on once we’re finished so that Asterisk-GUI doesn’t mangle the settings when you use Asterisk-GUI for other things. Of course, it also means you’ll need to log in as root to set up credentials for the "Incredible 9″ trunks, but that’s a walk in the park. In fact, that’s the beauty of using chattr in the first place.

For those that love to wade into the weeds of Asterisk design, there is another feature which permits storage of additional settings for any [context]. You simply create [context](+) in a separate file using the same name as the original context. When you reload Asterisk, it blends all of the entries from the two contexts. That’s exactly what we needed in order to simplify storage of credentials for providers using our own config files. Unfortunately, the Asterisk-GUI developers forgot about this syntax and removed our [context](+) entries presumably thinking they were bad code. This left the credentials themselves sitting in a config file with no context, and that wreaked all sorts of havoc in Asterisk-GUI. So now you know why we needed write protection for our credentials_sip.conf file.

Speaking of credentials, here’s how the default credentials_sip.conf file actually looks. To edit the file, you start by removing the immutable bit: chattr -i credentials_sip.conf. Never comment out the host entries or Asterisk dies! Host names may need to be changed depending upon the server on which your provider sets up your individual account. Everything else is simple enough for anyone to master without a tutorial. Plug in your ACCTNAME and ACCTPASS and uncomment the affected lines by removing the semicolons for any trunk you wish to use. Save your settings. Don’t forget to reprotect the file when you’re finished: chattr +i credentials_sip.conf. Finally, restart Asterisk: service asterisk restart. We plan to add scripts to automatically manage these trunk settings, but we wanted you to know how everything actually worked so you can do it yourself should you ever feel the urge.

[voipms](+)
; VoIP.ms trunk Prefix: Dial 9
;username = ACCTNAME
;secret = ACCTPASS
host = atlanta.voip.ms

[Vitelity](+)
; Vitelity trunk Prefix: Dial 8
;username = ACCTNAME
;secret = ACCTPASS
host = inbound1.vitelity.net

[lesnet_peer](+)
; Les.net trunk Prefix: Dial 7
;username = ACCTNAME
;secret = ACCTPASS
host = did.voip.les.net

[ipcomms](+)
; IPcomms trunk Prefix: Dial 6
;username = ACCTNUM
;fromuser = ACCTNUM
;secret = ACCTPASS
host = 2way.ipcomms.net

[didlogic](+)
; DIDlogic trunk Prefix: Dial 5
;username = ACCTNUM
;secret = ACCTPASS
host = sip.didlogic.net

[CallCentric](+)
; CallCentric trunk Prefix: Dial 4
;username = ACCTNUM
;fromuser = ACCTNUM
;authuser = ACCTNUM
;secret = ACCTPASS
host = callcentric.com

[FutureNine](+)
; FutureNine trunk Prefix: Dial 3
;username = ACCTNUM
;secret = ACCTPASS
host = incoming.future-nine.com

I hear some of you squawking, "Why do you call it ‘Incredible 9’ when there are only 7 providers?" The answer is that Google Voice is managed separately in credentials_googlevoice.conf because it operates differently in Asterisk. Anveo Direct also has a different way of handling outbound SIP calls. A PIN is required as part of the dial string. That PIN is managed separately in credentials_extensions.conf. So… 7 + 2 = 9.

Linux Application Framework for Incredible PBX

We also wanted to simplify the process of adding new Linux utilities to our Incredible PBX setup for Asterisk-GUI. You may know that Asterisk-GUI runs under a lean, mean web server that’s actually part of Asterisk. By default, it operates on port 8088. We wanted to leave it that way to simplify the procedure for compiling Asterisk to run as the asterisk user as opposed to the root user. The Asterisk web server never was intended to compete with Apache, and there is no support for PHP much less MySQL. In order to use the dozens of Incredible PBX utilities and databases as well as all of the text-to-speech and speech recognition tools, we needed Apache, PHP, and MySQL. So our design decision was to run Apache on port 80 with full PHP support and then run MySQL in the same way it has been installed on LAMP servers since Day One.

That design meant we still needed a separate web site to support Incredible PBX utilities. Luckily, our friends at Kennon Software built a beautiful user interface for PBX in a Flash many years ago. With some minor tweaking to account for newer releases of PHP, it was a perfect fit for Incredible PBX as you can see at the top of this article. It has all the things we were looking for including an RSS Feed to provide emergency announcements. It also provides developers unlimited flexibility to add local applications and make other modifications as desired for both end-users and administrators. So our tip of the hat again goes to Kennon Software for their terrific open source contribution to our projects.

Choosing a Linux Platform for Incredible PBX

Speaking of Linux, we’re often asked what’s the best Linux platform on which to run Incredible PBX. Our stock answer is ALL OF THE ABOVE. Incredible PBX has been and is being engineered to run well on almost any Linux platform. We plan to initially release Incredible PBX for Asterisk-GUI on the CentOS/Scientific Linux 6.5 platform, but we’ll add Ubuntu 14.04 and Debian in coming weeks. In this way, we can support all of our favorite low-cost hardware platforms including the Raspberry Pi, BeagleBone Black, CuBox, PogoPlug, and anything else we can get our hands on.

Adding Inbound & Outbound Dialplan Code to Incredible PBX

This week we also tackled some of the other items on the Wish List. We’ve heard from a number of folks that wanted a simple way to add customized dialplan code whenever a call was made or received. That was an easy one. In extensions_custom.conf, you’ll now find the following contexts which can be enhanced in any way you choose. Just plug your additional code into each context between the two default entries.

[incoming-sub]
exten => incoming-sub_1,1,Noop(*** Incoming: ${CALLERID(all)} on ${CHANNEL} ***)
exten => incoming-sub_1,n,Return()

[outgoing-sub]
exten => outgoing-sub_1,1,Noop(*** Calling: ${CALLERID(dnid)} from ${CALLERID(all)} ***)
exten => outgoing-sub_1,n,Return()

Managing Incoming Calls to "Incredible 9″ Trunks

We haven’t (yet) come up with a really simple way to adjust how inbound calls to our preconfigured trunks are processed because of the basic Asterisk-GUI design. In a nutshell, incoming calls come into your PBX on a phone number, aka DID. That DID is associated with a trunk that you’ve registered to a specific provider. Once the call hits Incredible PBX, we need to tell the PBX where to route the call. Typical choices include an extension, a group of extensions (i.e. a ring group), an IVR, or an AutoAttendant. There are others. In extensions.conf, you will find the dialplan code to manage incoming calls to the "Incredible 9″ trunks. The setup for each of the 9 trunks looks like this:

[DID_Vitelity]
include = DID_Vitelity_default
[DID_Vitelity_default]
exten = _.,1,Set(CALLERID(name)=${CALLERID(number)})
exten = _.,n,Set(CALLERID(number)=${CALLERID(number):0:10})
exten = _.,n,Gosub(cidlookup,cidlookup_1,1())
exten = _.,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten = _.,n,Gosub(incoming-sub,incoming-sub_1,1())
;exten = _.,n,Goto(default,6001,1)
;exten = _.,n,Goto(ringroups-custom-1,s,1)
;exten = _.,n,Goto(voicemenu-custom-2,s,1)
exten = _.,n,Goto(voicemenu-custom-1,s,1)
[CallingRule_OutVitelity]
exten = _8NXXNXXXXXX,1,Macro(trunkdial-failover-0.4,${Vitelity}/${EXTEN:1},,Vitelity,)
exten = _81NXXNXXXXXX,1,Macro(trunkdial-failover-0.4,${Vitelity}/${EXTEN:2},,Vitelity,)

In the Asterisk-GUI world, a [DID_provider] context manages incoming calls FROM a provider’s DID or trunk. And a [CallingRule_provider] context manages outbound calls TO your provider’s trunk. In the default Vitelity setup, incoming Vitelity calls get routed to voicemenu-custom-1, and outbound calls with an 8 prefix get routed out through your Vitelity trunk. As noted, all of these contexts can be found in extensions.conf.

Toward the end of the default context you will see two voicemenu-custom entries as well as a ringroups-custom-1 entry and a default entry with an extension number. All but one of these is commented out. As you have probably guessed, the uncommented entry determines where the incoming call is routed. When you create IVRs or ring groups in Asterisk-GUI, each new creation gets assigned a sequential number starting with 1. ringroups-custom-1 is a preconfigured Ring Group that currently sends calls to extensions 6001 and 6002, the two extensions created in the Incredible PBX default setup. If you add additional extensions and then add those new numbers to this preconfigured ring group, then those phones will ring as well. It does NOT change the sequential number originally assigned to this ring group. Adding a new ring group does that.

In Asterisk-GUI, IVRs and AutoAttendants are called Voice Menus. Incredible PBX ships with two. Voice Menu #1 is the Nerd Vittles’ Stealth AutoAttendant. It greets the caller with a cheery message from Allison while providing a couple seconds for someone (like you) to press a button to reroute the call to an undisclosed destination. If no key is pressed, the incoming call is routed to Ring Group #1.

Voice Menu #2 is a Demo IVR that showcases many of the Nerd Vittles applications. By default, all "Incredible 9″ trunks are configured to route incoming callers through the Stealth AutoAttendant to Ring Group #1. If you’d prefer to route incoming calls to a ring group or a particular extension or the demo IVR, the commented out entries will let you do that.

But suppose you wanted to route an incoming call to a custom extension defined in extensions_custom.conf? Well, it’s easy. Just change the context to CallingRule_extensions_custom and route the call to line 1 of the extension context desired. For example, to send an incoming call to the AsteriDex Voice Dialer (411) which lets callers say the name of the party they wish to reach, you’d insert a call destination entry that looked like this:

exten = _.,n,Goto(CallingRule_extensions_custom,411,1)

Better yet, you can use the generic dialplan context, DLPN_DialPlanMain, to reach any extension on your server:

exten = _.,n,Goto(DLPN_DialPlanMain,411,1)

As you add new ring groups, extensions, and voice menus with Asterisk-GUI, you can adjust these settings accordingly now that you know how all of this works. After making changes in extensions.conf, be sure you’ve only enabled ONE destination per trunk by commenting out the rest of them. Then reload your dialplan: asterisk -rx "dialplan reload"

Free Worldwide Calling Support with Incredible PBX

We mentioned last week that iNum support will be included through two SIP providers to let you make free phone calls worldwide to anyone with a registered iNum. SIP URIs are equally important for the same reason. You can make a free call to anyone, anywhere in the world if the recipient happens to have a SIP URI, and sip2sip.info will provide a free one to anybody. To support SIP URIs, Incredible PBX for Asterisk-GUI includes a new context that will let you link a SIP URI to an extension on your PBX. We’ve included an entry for L-E-N-N-Y to get you started. You can add as many more as you like:

[CallingRule_SIP_URI]
exten = 53669,1,Dial(SIP/2233435945@sip2sip.info)

CallerID Name Lookups with Incredible PBX

On the CallerID Name front, we’re still exploring alternatives including incorporation of CallerID Superfecta which originally was a Nerd Vittles creation. It now is maintained by the POSSA Development Team. In the interim, we’ve provided code for one of the best CNAM sources in the business, OpenCNAM. It gets you 10 free lookups an hour from cached entries. If you need more, you can sign up for an account. For completed calls, there is a charge of $.004. Just adjust the CURL entry below to plug in your credentials:

[cidlookup]
exten => cidlookup_1,1,Set(CURLOPT(httptimeout)=7)
exten => cidlookup_1,n,Set(CALLERID(name)=${CURL(https://account_sid:auth_token@api.opencnam.com/v2/phone/${CALLERID(num)}?format=pbx&ref=incrediblepbx)})
exten => cidlookup_1,n,Set(current_hour=${STRFTIME(,,%Y-%m-%d %H)})
exten => cidlookup_1,n,Set(last_query_hour=${DB(cidlookup/opencnam_last_query_hour)})
exten => cidlookup_1,n,Set(total_hourly_queries=${DB(cidlookup/opencnam_total_hourly_queries)})
exten => cidlookup_1,n,ExecIf($["${last_query_hour}" != "${current_hour}"]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_1,n,ExecIf($["${total_hourly_queries}" = ""]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_1,n,Set(DB(cidlookup/opencnam_total_hourly_queries)=${MATH(${DB(cidlookup/opencnam_total_hourly_queries)}+1,i)})
exten => cidlookup_1,n,ExecIf($[${DB(cidlookup/opencnam_total_hourly_queries)} >= 60]?System(${ASTVARLIBDIR}/bin/opencnam-alert.php))
exten => cidlookup_1,n,Set(DB(cidlookup/opencnam_last_query_hour)=${current_hour})
exten => cidlookup_1,n,Return()
exten => cidlookup_return,1,ExecIf($["${DB(cidname/${CALLERID(num)})}" != ""]?Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}))
exten => cidlookup_return,n,Return()
;--== end of [cidlookup] ==--;;end of Incredible PBX original build file

Managing Outbound CallerID & PINs with Incredible PBX

There appear to be a few leftover CallerID bugs in the original Asterisk-GUI code. As a workaround, we’ve added support for setting CallerID numbers in credentials_extensions.conf. This lets you set outbound CallerID numbers on a per trunk basis for providers that allow spoofing of CallerID numbers.

Here’s what the complete credentials_extension.conf file actually looks like. The AnveoPIN is the code used to authorize outbound calls through Anveo Direct. Anveo handles outbound calling differently than most providers so the setting had to go here instead of in the more traditional credentials_sip.conf file. Sorry. The other entries are self-explanatory. This config file can be edited using your favorite text editor. Then service asterisk restart and you’re done.

AnveoPIN = 024680
; Conference Bridge PINs
CONF_USER_PIN = 1234
CONF_ADMIN_PIN = 4321
; DISA password
DISA_PW = 9876
; CallerID numbers
CID_allroutes = 8005551212
CID_CallCentric = 8005551212
CID_ipcomms = 8005551212
CID_voipms = 8005551212
CID_anveodirect = 8005551212

Outbound Call Processing with Incredible PBX

With apologies to our international friends, we’ve included a template to handle processing of all outbound U.S. calls using the "Incredible 9″ trunks. Basically, you can dial a 10-digit number or 1+ the 10-digit number, and the [outbound-allroutes] context will walk the call through all of the trunks for which you’ve registered including Google Voice. If a trunk isn’t registered, it’s skipped. The arrangement of the trunks can be adjusted to meet your own needs. As delivered, calls are processed in the following order: Google Voice, VoIP.ms, Vitelity, les.net, IPcomms, DIDlogic, CallCentric, FutureNine, and Anveo Direct. You obviously can add as many additional providers as desired or rearrange the ones that already are included. And international calling can be added easily using the existing entries as a model. Cut-and-paste is your friend!

[outbound-allroutes]
exten => _NXXNXXXXXX,1,Set(CALLERID(num)=${CID_allroutes})
exten => _NXXNXXXXXX,n,Dial(Motif/GoogleVoice/1${EXTEN}@voice.google.com)
exten => _NXXNXXXXXX,n,Dial(${voipms}/${EXTEN})
exten => _NXXNXXXXXX,n,Dial(${Vitelity}/${EXTEN})
exten => _NXXNXXXXXX,n,Dial(${lesnet_peer}/1${EXTEN})
exten => _NXXNXXXXXX,n,Dial(${ipcomms}/${EXTEN})
exten => _NXXNXXXXXX,n,Dial(${didlogic}/1${EXTEN})
exten => _NXXNXXXXXX,n,Dial(${CallCentric}/1${EXTEN})
exten => _NXXNXXXXXX,n,Dial(${FutureNine}/1${EXTEN})
exten => _NXXNXXXXXX,n,Dial(SIP/${AnveoPIN}1${EXTEN}@sbc.anveo.com)
exten => _NXXNXXXXXX,n,Hangup
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=${CID_allroutes})
exten => _1NXXNXXXXXX,n,Dial(Motif/GoogleVoice/${EXTEN}@voice.google.com)
exten => _1NXXNXXXXXX,n,Dial(${voipms}/${EXTEN:1})
exten => _1NXXNXXXXXX,n,Dial(${Vitelity}/${EXTEN:1})
exten => _1NXXNXXXXXX,n,Dial(${lesnet_peer}/${EXTEN})
exten => _1NXXNXXXXXX,n,Dial(${ipcomms}/${EXTEN:1})
exten => _1NXXNXXXXXX,n,Dial(${didlogic}/${EXTEN})
exten => _1NXXNXXXXXX,n,Dial(${CallCentric}/${EXTEN})
exten => _1NXXNXXXXXX,n,Dial(${FutureNine}/${EXTEN})
exten => _1NXXNXXXXXX,n,Dial(SIP/${AnveoPIN}${EXTEN}@sbc.anveo.com)
exten => _1NXXNXXXXXX,n,Hangup

DISA Support for Incredible PBX

Unless we missed it, Asterisk-GUI was missing DISA support, the ability to call your PBX and receive dialtone to make an outbound call through the PBX. Because of costs associated with outbound calls, this can make a real difference in some countries. We’ve added DISA support through the D-I-S-A (3472) extension. The DISA password can be set in credentials_extensions.conf. The DISA extension can be added to an IVR for one or more trunks to provide password-protected DISA call access to incoming callers. The dialplan code can be adjusted to meet your own requirements. As delivered, only 10-digit calls are permitted. Just change the 10 on line 10 if you want to enable international dialing. Calls are limited to 150 minutes by default. Just change the 9000 (seconds) entry as desired.

[custom-disa]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(TIMEOUT(digit)=7)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(enter-password)
exten => s,n,Read(MYCODE,beep,7)
exten => s,n,Noop(DISA_PW: ${DISA_PW})
exten => s,n,GotoIf($["${MYCODE}" = "${DISA_PW}"]?disago:bad,1)
exten => s,n(disago),Set(TIMEOUT(absolute)=9000)
exten => s,n,Read(NUM2CALL,pls-entr-num-uwish2-call,10)
exten => s,n,Background(calling)
exten => s,n,SayDigits("${NUM2CALL}")
exten => s,n,Goto(outbound-allroutes,${NUM2CALL},1)
exten => s,n,Hangup
exten => t,1,Hangup
exten => i,1,Hangup
exten => h,1,Hangup
exten => bad,1,Hangup

Call Forwarding Support with Incredible PBX

We have restored the call forwarding functionality that originally was missing in Asterisk-GUI. The same feature codes found in FreePBX are supported. By dialing *72, you can set forwarding for any extension to either a local number or any other number supported by your dial plan. By dialing *72NXXNXXXXXX or *726XXX (local extensions typically are in the 6000-6299 range with Asterisk-GUI), you can set call forwarding in a single step. *73 can be used to disable call forwarding for the extension from which you dialed, or *74 can be used to disable call forwarding for any extension on your server. A list of currently forwarded extensions can be retrieved using the Asterisk CLI: asterisk -rx "database show CF"

Conference Bridge Support in Incredible PBX

As previously mentioned, Conference Bridge support wasn’t available when Asterisk-GUI was released so we’ve added it. Just dial C-O-N-F (2663) to join the conference bridge. User and admin PINs are set in the credentials_extensions.conf file. You can create as many of these as you need by cloning the code below with different extension numbers:

[conf_bridge]
exten => 2663,1,Macro(user-callerid,)
exten => 2663,n,Set(MEETME_ROOMNUM=2663)
exten => 2663,n,Set(MAX_PARTICIPANTS=0)
exten => 2663,n,Set(MEETME_MUSIC=default)
exten => 2663,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?READTHEPIN)
exten => 2663,n,Answer
exten => 2663,n,Wait(1)
exten => 2663,n,Set(PINTRIES=0)
exten => 2663,n,Noop(${CONF_USER_PIN})
exten => 2663,n(READTHEPIN),Read(PIN,enter-conf-pin-number,,,,)
exten => 2663,n,GotoIf($[${PIN} = ${CONF_USER_PIN}]?ENDUSER)
exten => 2663,n,GotoIf($[${PIN} = ${CONF_ADMIN_PIN}]?ADMINISTRATOR)
exten => 2663,n,Set(PINTRIES=$[${PINTRIES}+1])
exten => 2663,n,GotoIf($[${PINTRIES}>3]?h,1)
exten => 2663,n,Playback(conf-invalidpin)
exten => 2663,n,Goto(READTHEPIN)
exten => 2663,n(ADMINISTRATOR),Set(CONFBRIDGE(user,admin)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,marked)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,dsp_drop_silence)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,talk_detection_events)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,announce_user_count)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,announce_join_leave)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,music_on_hold_when_empty)=yes)
exten => 2663,n,Goto(ext-meetme,STARTMEETME,1)
exten => 2663,n(ENDUSER),Noop(User Options:)
exten => 2663,n,Set(CONFBRIDGE(user,dsp_drop_silence)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,talk_detection_events)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,announce_user_count)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,announce_join_leave)=yes)
exten => 2663,n,Set(CONFBRIDGE(user,music_on_hold_when_empty)=yes)
exten => 2663,n,Goto(ext-meetme,STARTMEETME,1)
exten => 2663,hint,confbridge:2663

Google Voice Support in Incredible PBX

Google Voice is another little goodie that wasn’t available when Asterisk-GUI came along. Because we’re now running everything on the Asterisk 11 platform, it seemed silly not to include Google Voice support. And we’ve made it about as easy to set up as tying your shoes. Plug in your Google email address and password using the new web interface, and you’re done. By default, all 10-digit and 11-digit outbound calls are first attempted through your Google Voice trunk. You can’t beat free!

If you’ve used the same account elsewhere, Google may block access from your new IP address. In this case, just follow the steps outlined in this Google Reset Procedure to get things going.

Managing Call Detail Records (CDR) in Incredible PBX

The gorgeous CDR Viewer found under the Options -> Advanced Options -> Enable tab in Asterisk-GUI is incredibly flexible. In addition to being lightening fast, you can reorder the CDR listing by simply clicking on any column heading. Clicking twice will sort the list in the opposite order. You also can expand the detail in two ways. Either click on an individual entry (as shown) to display the complete CDR entry or check the Show All Fields checkbox to get the full picture for every CDR entry. The complete CDR database in CSV format can be retrieved from /var/log/asterisk/cdr-csv/Master.csv.

Getting Up to Speed on Asterisk-GUI Basics

We’ve covered a lot of territory this week. You don’t have to master it all at once. Incredible PBX is being engineered to give you the best of both worlds rather than one size fits all. By setting things up this way, you can add your own features and share them with the community as you move up the learning curve. That’s what open source is all about!

The other goal was to leave Asterisk-GUI intact to the greatest extent possible. This has several advantages. First, for previous users of Asterisk-GUI, they’ll feel right at home. Second, we don’t have to write extensive documentation for Asterisk-GUI because many others have already done the heavy lifting. One obvious word of caution. Don’t delete, rename, or otherwise modify the default trunks, users, calling rules, ring groups, and dialplans that already have been created to support Incredible PBX. If you do, you will break things. But feel free to add as many new pieces to your setup as desired and, of course, the extension passwords can be changed in any way you like. Trunk credentials for the "Incredible 9″ preconfigured trunks should be managed using the credentials files documented above.

Here are a few resources that will guide you through mastering the Asterisk-GUI:

Remember the Objective of Incredible PBX for Asterisk-GUI

We’ll close for today by reiterating why we’re introducing a new VoIP alternative with Incredible PBX for Asterisk-GUI. From the ground up, this project is designed as an open source, hobbyist platform so that you can actually LEARN how Asterisk works and become self-sufficient in designing AND managing your own VoIP communications platform. Is it "Pure GUI"? Nope. Here’s why. One of the major reasons that so many folks have had their VoIP systems hacked over the past few years is because those users never quite understood how their "Pure GUI" stuff was working (or not) under the hood.

Does "hobbyist platform" mean it’s a Crappy Purple Scion? Nope. In fact, the Asterisk-GUI tools and Asterisk code in Incredible PBX were designed and written by some of the best Asterisk experts in the business including Mark Spencer, the creator of Asterisk. When the nay-sayers snicker at your "hobbyist platform," just smile and enjoy your independence. 🙂

Originally published: Monday, December 8, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…

Incredible PBX on Steroids: The Asterisk-GUI Pilgrimage Begins (Chapter 1)

As the holiday season gets underway with Thanksgiving, Hanukkah, Christmas, and especially Festivus, we thought it might be interesting to actually provide a running dialog of how a new Asterisk® project is born and what hurdles and solutions are encountered along the way. We mentioned last week that we were dusting off Mark Spencer’s Asterisk-GUI with hopes of transforming it into an updated Asterisk 11 platform for hobbyists and SOHO telephony users with many of the ease-of-use touches that have made Incredible PBX a big hit. So today we officially kick off the adventure with a look back at Week One. Our target, by the way, is a New Year’s Day release to celebrate the arrival of 2015.

This is the first installment in our series. You can catch up with the Overview as well as Chapter 2 and Chapter 3 here.

Project Development Roadmap

You may be asking, "What’s in it for me?" Well, lots! One of the unfortunate side effects of having always relied upon the FreePBX® GUI for Asterisk administration is you never really learned how Asterisk works. Nor did we ever quite appreciate its lightning-fast performance. We’re as guilty as anyone for over-reliance on a design tool without much appreciation for its interaction with the actual communications server. And, like many things in life, you form some bad habits along the way that are hard to break. Don’t get us wrong. There are thousands of things to like about FreePBX and, for production-level servers hosting dozens or hundreds of users, it remains a very comfortable choice and our hands-down favorite.

We resolved early on to approach the Asterisk-GUI remake a little differently. We plan to actually document why we’re going down certain paths and what the benefits will be for the ultimate user. There won’t be any convoluted code to deter your learning how things actually work. And there won’t be any patent, trademark, or copyright gotchas to hinder your forking or repurposing our code to meet your own requirements. And, finally, there won’t be any license fees, hidden or otherwise. Just comply with the GPL2 license as written and be our guest! From our vantage point, that’s what open source is all about.

Defining Project Objectives

We began the week by sketching out some objectives as well as defining some likes and dislikes. As we mentioned last week, the objective is not to replace FreePBX for those that actually need that horsepower. First and foremost we want to design this product for the target audience: hobbyists, home users, and SOHO businesses. Many of the platforms we are targeting have limited memory and only modest computational ability. Many of the people in the target audience have never used a PBX before and know little to nothing about networks and security. We don’t want anyone blindsided by a $100,000 phone bill because they didn’t know how to implement a firewall so we’ll include a preconfigured one as part of the install. And, like all Incredible PBX systems, an automatic update utility will be included to keep your system current AND safe!

Second, we wanted a product that was incredibly simple to put into production. Ease of configuration was a definite must-have. With many GUIs (think: Microsoft Windows), developers get so enamored with the brilliance of their own creation that they lose sight of the fact that typing a short list of usernames and passwords often is much simpler than navigating through dozens of data entry screens with hundreds of mouse clicks to enter the same information.

We also are steering clear of reinventing the proverbial wheel. Mark Spencer and his colleagues are some of the most talented programmers on the planet. To the extent that the original, feature-rich Asterisk-GUI creation can be implemented without major plumbing changes, that is not only desirable but absolutely essential in bringing this new product to market within weeks, not months or years.

Keep in mind that both FreePBX and Asterisk-GUI are code generators for Asterisk. No call is actually processed by FreePBX or Asterisk-GUI. From a system design standpoint, we wanted Asterisk to be self-sufficient on this new Incredible PBX platform. Stated another way, we didn’t want Asterisk to fail just because Apache or MySQL had system failures since neither of them is required for Asterisk to function reliably in the first place. It’s one thing for your GUI or MySQL database to be inoperable. It’s quite another when it also brings down your entire phone system.

In summary, we are lifelong believers in the KISS principle. Keep It Simple, Stupid. As much as we love FreePBX, its system design is anything but simple. Configuration information is embedded in hundreds of HTML files, Linux templates, Asterisk configuration files including AstDB plus 100+ MySQL tables. By contrast, Asterisk-GUI uses a tiny collection of native Asterisk .conf files to configure virtually all its settings. We wanted to preserve that "pure Asterisk" simplicity.

One of the other real advantages of the Asterisk-GUI design is you can create something in the GUI and then review the Asterisk-generated code in /etc/asterisk to see exactly how the original Asterisk developers intended the feature to work. In addition to the learning experience, it makes it easy to debug coding errors and to make adjustments and customizations to meet individual needs without inadvertently bringing down the whole house of cards.

We wanted a product that was easy for an administrator to maintain, to update, AND to back up. After all, this is a phone system not a rocketship. It shouldn’t take a rocket scientist to maintain it. And it won’t.

Project Design 101: Preconfigured Trunks, Extensions & Routes

With these objectives in mind, we’ve made some design choices on the front end that are worth mentioning. Configuration settings for SIP, IAX, and Google Voice trunks give new users more headaches than any other single feature in a new PBX. So we’re taking much of the pain out of that process by providing 9 preconfigured trunks. Meet the Incredible 9: Google Voice, Vitelity, VoIP.ms, Les.net, IPcomms, DIDlogic, CallCentric, FutureNine, and Anveo Direct. Outbound calling is managed by routes that are tied to individual extensions. These can be adjusted quickly in the GUI. We’ve chosen to set up outbound calling for the Incredible 9 using preconfigured dialing prefixes. No prefix or a 1-prefix sends the call out through Google Voice and, if Google Voice isn’t available, then the call is routed through the next working outbound trunk in the order shown above. A prefix of 2-9 sends the call out through one of the preconfigured trunks. We’ve also included support for free worldwide iNum calling using either VoIP.ms or CallCentric. Both vendors will also provide you with a free iNum DID. Just dial your iNum prefix of 0 (CallCentric) or 90 (VoIP.ms) followed by the last 7 digits of any assigned iNum DID to place a free call. As usual, Lenny stands ready to provide 24/7 technical support through his iNum DID: And, of course, all of these settings can be modified or tweaked to your liking using Asterisk-GUI!

A word about the "Incredible 9″ providers. The major prerequisite for inclusion was communications compatibility with Asterisk without any firewall exposure of Asterisk ports. That means the provider had to support outbound and/or inbound calling without any port exposure of Asterisk to the Internet. Vitelity and Google have been major financial supporters of our projects over the years so they made the short list. Both also offer incredible pricing and feature-rich VoIP implementations. The others made the cut based upon great user satisfaction reports, free services of one type or another, or dirt cheap pricing. Can you add additional providers using Asterisk-GUI? Absolutely. But the "Incredible 9″ each can be activated in under 10 seconds after you’ve signed up for an account with your choice of providers. In the VoIP world, there’s little reason not to choose several since you only pay for the services you actually use, and we would encourage you to do so.

Incoming call processing also is preconfigured with some extensions, a ring group, a Stealth AutoAttendant, DISA, and an IVR with an assortment of Incredible PBX applications for Asterisk. All can be modified or embellished to meet your own requirements.

Bottom Line: You get a turnkey PBX that’s ready to go. It’s also easily configurable to meet your most demanding requirements. Incredible PBX delivers The Best of Both Worlds using native Asterisk code.

A Fresh Look at Managing Credentials

One of the more exasperating realities of password management with FreePBX is the number of places you have to look to find or change passwords. Some are stored in various Asterisk .conf files. Voicemail passwords are hidden away in text strings in voicemail.conf. Others are stored in MySQL tables. Some are encrypted, and some aren’t. Asterisk-GUI took a different approach and stores all passwords in the Asterisk .conf files in /etc/asterisk.

As talented as the FreePBX and Asterisk-GUI programmers are, we don’t trust any web-based application to remain secure if it’s directly exposed to the Internet. If you do, you’re either nuts or have plenty of money to burn. GUIs should be reserved for administrator use behind a secure firewall, period. In our new design, you need firewall whitelist privileges plus root or asterisk user privileges plus GUI admin user access to gain access to passwords. If all of these layers are compromised, passwords are the least of your worries.

We’ve taken password management one step further. As best we can given the design choices in Asterisk 11 and Asterisk-GUI, we’ve aggregated as many passwords as possible into new credentials config files: credentials-sip.conf, credentials-googlevoice.conf, and credentials-extensions.conf. There’s one for the "Incredible 9″ SIP providers. There’s one for Google Voice. And there’s a catchall for various passwords, PINs, and predefined CallerID numbers for various trunks. These are straight-forward text files that can be quickly edited using any text editor. Plug in your account names, passwords, and PINs. Optionally, adjust the providers’ server addresses as required. And you’re done. If you can tie your shoes, you can do this. Quick and functional, not fancy!

Redesigned Conferencing Solution for Asterisk 11

MeetMe conferencing as originally implemented in Asterisk-GUI required an external timing source. This timing source was provided by analog boards on some of the commercial hardware platforms on which Asterisk-GUI was deployed. For our target audience, we’re assuming that most people probably want to ditch Ma Bell and costly landlines as part of the migration to a new PBX platform. So, even though Asterisk-GUI still supports analog trunks, we have chosen to offer the Asterisk 11 Conference Bridge option which does not require an external timing source. The new Conference Bridge is preconfigured out of the box. Set up user and admin PINs. And you’re done. Dial C-O-N-F (3663) to join the conference.

The Baker’s Dozen Incredible PBX Apps: Alive and Well

We closed out Week One with some minor tweaking of several of our favorite Incredible PBX applications to accommodate the new Asterisk-GUI platform. We’re pleased to report that everything still works. Because of changes imposed by Google, you’ll need to jump through a few hoops to implement Speech Recognition support on this new Asterisk platform. All of the necessary software has already been put in place so all you need is an API key from Google. Once you obtain it, simply plug it into line 70 of speech-recog.agi. No other configuration is required. The affected applications are marked with an asterisk (*) below. But the good news is, if you’ve used these Nerd Vittles applications in the past, you’ll feel right at home.

Stay tuned for more and… HAPPY THANKSGIVING!

Continue reading Chapter 2

Originally published: Monday, November 24, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…

VoIP Navigation Guide: Getting Started with Asterisk and FreePBX


When you were just getting started with Asterisk® in the early days, you had two choices: hire a consultant to build you an Asterisk system or start with Asterisk@Home and learn it yourself. That was a disaster for many folks. Times have changed, and there are literally dozens of aggregations and platforms from which to choose. But the question we continue to hear is "What’s the best way to get started?" Today’s VoIP Navigation Guide will help you make the right choices.

Before we begin, you need to do a little head-scratching yourself. Sit down with a pencil and paper (or a computer if you must) and jot down answers to our Top 10 Preliminary Questions:

  1. Is this for home or office use?
  2. How many simultaneous calls?
  3. How many users on the system?
  4. Will there be remote or traveling users?
  5. Is this a mission-critical system for you/others?
  6. What type & speed Internet service? Wi-Fi only?
  7. What is the skillset of those supporting the system?
  8. Do you want to babysit hardware for your system?
  9. What’s your initial and monthly budget for the project?
  10. What should happen to calls if your house/office burns down?

Skillset Matters! Let’s start with the obvious. The technical skillset of you and any other people that will be managing your VoIP server are critically important. This isn’t the old days where you only had to monitor people making long distance calls from within your own house. Once you connect a VoIP server to the Internet, anybody and everybody around the world can take a shot at your server and run up huge phone bills on your nickel unless you know what you’re doing or unless you deploy a server on which access is locked down to just you and trusted users and service providers.

We preach (regularly) that firewalls are essential if you’re going to deploy a VoIP server. In the home or office environment, that means that, in addition to your VoIP server, you also need a hardware-based firewall/router with no mapped ports to the VoIP server, period. Any other setup and it’s just a matter of time until you’re hacked.

In the hosted or cloud environment, it means at the very least a software-based firewall on your VoIP server with all access restricted to a whitelist of trusted users and providers. Any other setup and it’s just a matter of time until you’re hacked.

If you’re not qualified to manage either a hardware or software firewall, then your VoIP choices are limited. None of the major aggregations including PBX in a Flash, the FreePBX® Distro, AsteriskNOW, and Elastix provide any firewall protection as installed. While Fail2Ban is included, it is basically a log scanner which searches for failed login attempts and blocks IP addresses that make excessive login attempts. The major problem with Fail2Ban is that it takes time to run and, if your server is attacked from powerful servers, that may not happen until thousands of hack attempts have been executed.

We have attempted to address this problem with this summer’s new releases of Incredible PBX. In these new releases, whitelist access is locked down as part of the installation process. You have a choice of platforms.

On Cloud-based servers and depending upon your installation skills, we recommend:

On self-managed servers, you typically install the Linux operating system and then run the Incredible PBX installer. On smaller devices, we handle that for you. We recommend the following setups with the caveat that the old adage still applies: "You get what you pay for!" All four of the small hardware offerings below support WiFi-only operation. Just add the recommended WiFi USB dongle. For the CuBox-i, it’s built in. The VirtualBox setup takes less than 10 minutes.

Sizing Your Platform. Appropriate server and Internet capacity obviously turns on most of the answers you wrote down in the preliminary questionnaire. If the system will be used by less than a handful of people, you’re probably safe with the cloud-based solutions we’ve identified or one of the four low-cost devices listed above. Keep in mind that you need roughly 100Kbps of Internet bandwidth for each simultaneous VoIP call. If you have existing POTS lines from Ma Bell, those don’t consume Internet bandwidth but do consume local network resources. POTS line integration also requires additional hardware for each line. For less than 5 POTS lines, the OBi110 is an excellent choice. You’ll find it advertised in the right column of Nerd Vittles for under $50.

For up to a couple dozen low-call-volume employees, the RentPBX Cloud offering is a terrific bargain. It includes the necessary bandwidth not only to make calls but also to connect your extensions. When you get above those numbers of users or with heavy call volume, scaling matters. You don’t want to purchase a server only to discover on Day Two that it can’t handle the call volume. Here’s where the PBX in a Flash Forum can be a tremendous help. Describe your environment using the Top 10 Checklist from above. One of our hundreds of experts will lend a hand in recommending what you need to get started. Better yet, hire one of the gurus to handle the setup for you. It’ll save you thousands of dollars in headaches and easily pay for itself in future savings.

The PBX in a Flash Alternative. We haven’t mentioned PBX in a Flash as a solution for those just beginning their VoIP adventure. The reason is simple. The firewall is not preconfigured on PBX in a Flash, and somebody has got to do it unless your server is sitting behind a rock-solid, hardware-based firewall. The beauty of PBX in a Flash is that it’s incredibly flexible. You can choose not only the version of Asterisk and FreePBX to install, but you also can compile Asterisk with any collection of features desired. Once you get your feet wet with Incredible PBX, it’s our VoIP tool of choice, but it takes some skills on your part to run it safely. A good place to begin is the Nerd Vittles Quickstart Guide for PBX in a Flash 3. Enjoy!

Originally published: Wednesday, September 17, 2014


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! Please have a look and post your support questions there. Our forum is extremely friendly and is supported by literally hundreds of Asterisk gurus.



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…

FMC: The Future of Telephony with Vitelity’s vMobile and Asterisk in the Cloud




If making phone calls from a web browser is what you’ve always longed for, then you’re in good company with Google and its future direction in the telephony space. Call us old fashioned but this strikes us as a solution in desperate need of a problem. What’s wrong with a Plain Old Telephone or a smartphone for making connections with friends and business associates? The real head scratcher is the fact that the WebRTC and Hangouts push demonstrates that the wizards at Google are seriously out of touch with the next generation. Will our 14-year-old daughter use Skype or Hangouts or FaceTime? Sure. About once a month to chat with Grandma or to interact with cousins scattered around the country, it’s a terrific option. And the same is true in the business community. When you need to collaborate with a half dozen colleagues, conferencing applications are invaluable. But to meet 95% of day in and day out business requirements, a telephone or smartphone is the clear device of choice. So join us today in celebrating the end of Google Voice XMPP service and the beginning of a new and even more exciting VoIP era… sans Google.


Of course, if it were up to the next generation, telephone calls might completely disappear in favor of text messaging, Snapchat, Instagram, and any other platform that includes recorded photos or videos. Note the subtle difference. Kids really are not interested in live video interaction. They find posed images that tell a story much more appealing. Why? Because recorded photos and videos let users present their best face, their movie star pose, and their expression of what they want others to perceive they’re really like. In short, live video is too much like real life. Our conclusion for those targeting the next generation is you’d better come up with something better and quite different than Skype, Hangouts, and FaceTime.

It’s Fixed-Mobile Convergence, Stupid!

Now let’s return to our primary focus for today, the current business community. Suffice it to say, there are a dwindling number of what we used to call "desk jobs" where an employee arrives at his or her desk at 9 a.m. and leaves at 5 p.m. As more and more jobs are headed off shore, the telephone and smartphone have replaced the corporate desk as the most indispensable corporate fixture. Particularly in the American marketplace, what we see with most businesses is a management layer and an (upwardly) mobile force of salespeople, consultants, and implementers that interact primarily through PBXs in an office headquarters or home office together with smartphones for those that generally are on the road. Many of these Road Warriors don’t even have a home phone any longer.


The telephony Holy Grail for this new business model is Fixed-Mobile Convergence (FMC). It’s the ability to transparently move from place to place while retaining your corporate identity. Every employee from the night watchman in Miami to the salesperson making calls from a Starbucks in California to the CEO in New York has an extension on a PBX in the cloud together with the ability to accept and place calls using the company’s CallerID name and number, transfer calls, and participate in conference calls regardless of whether the phone instrument happens to be a desktop phone or a smartphone. Is this even possible? Well, as of last week, the answer is ABSOLUTELY.

Vitelity has been a long-time corporate sponsor of both the Nerd Vittles and PBX in a Flash open source projects so we were thrilled when we were offered a free, Samsung Galaxy S III to try out the new (live) vMobile service that took Best in Show honors at ITEXPO Miami in January. As Vitelity’s Chris Brown would probably tell you, it’s one thing to demonstrate a new technology at a trade show and quite another to bring it into production. But Vitelity did it:



What we want to stress up front is that we’ve received no special treatment in getting this to work. We received the phone, opened a support ticket to register the phone on Vitelity’s vMobile network, and plugged our new credentials into the phone so that it could be integrated into our PBX in a Flash server. Once the smartphone became an extension on our PBX, we could place calls through our PBX with the S3 using both WiFi and Sprint 3G/4G service. Switching between WiFi and cellular is totally transparent. The CallerID for all outbound calls was our standard PBX CallerID. We also could place calls to other extensions on the PBX by dialing a 4-digit extension while connected to WiFi or the Sprint network virtually anywhere. If you have 3-digit extensions, those are a problem over the Sprint network but we’ll show you a little trick to get them working as well.

Keep in mind that every call from the S3 goes out through the PBX just as if you were using a standard desktop phone as a hardwired extension. And it really doesn’t matter whether the S3 has a WiFi connection or a pure cellular connection on Sprint’s network. You receive calls on the S3 in much the same way. It’s just another extension on your PBX. If you want to add it to a ring group to process incoming calls, that works. If other users on your PBX wish to call the S3 directly using the extension number, that works as well. If you want to transfer a call, pressing ## on the S3 initiates the transfer just as if you were using a phone on your desk. When we say transparent convergence, we really do mean transparent. No recipient of a call from the vMobile S3 would have any idea whether you were sitting at a desk in the corporate headquarters in New York or in a seat on a Delta jet after landing in San Francisco. Both the call quality and the corporate CallerID would be identical. And your secretary on maternity leave at Grandma’s house still could reach you using her vMobile S3 by simply dialing your corporate extension.

So that’s the Fortune 500 view of the new VoIP universe. How about the little guy with a $15 a month PBX in a Flash server in the RentPBX cloud1, a couple mobile sales people, and a handful of construction workers that build swimming pools for a living? It works identically. Each has an S3 connected as an extension on the PIAF cloud server. And calls can be managed in exactly the same way they would be handled if everyone were sitting side-by-side at desks in an office headquarters somewhere. The silver lining of cloud computing is that it serves as the Great Equalizer between SOHO businesses and Fortune 500 companies. Asterisk® paired with inexpensive cloud hosting services such as RentPBX lets you mimic the Big Boys for pennies on the dollar. We think Vitelity has hit a bases loaded, home run with vMobile.


vMobile Pricing

We know what you’re thinking. "Since you got yours for free, what does it really cost??" The Galaxy S3 (or S4) is proprietary running Trebuchet 1.0, a (rooted) CyanogenMod version of Android’s KitKat. You can purchase these devices directly from the Vitelity Store. Currently, you can’t bring your own device. The refurbished S3 is $189 including warranty. Works perfectly! That’s what we’re using. Next, you’ll need a vMobile account for each phone. Unless you’re a Nerd Vittles reader, it’s $9.95 per month. That gets you free WiFi calling and data usage anywhere you can find an available WiFi hotspot. And text messaging is free. For calls and data using Sprint’s nationwide network, the calls are 2¢ a minute and the data is 2¢ per megabyte ($20 per gigabyte). For us, a typical day of data usage with an email account and light web use costs about a quarter. YMMV! So long as you configure Android to download application updates when connected to WiFi, data usage should not be a problem unless you’re into photos and streaming video. Android includes excellent tools for monitoring and even curbing your data usage if this is a concern.

vMobile Gotchas

Before we walk you through the setup process, let’s cover the gotchas. The list is short. First, we don’t recommend connecting vMobile devices to a PBX sitting behind a NAT-based firewall, or you may end up with some calls missing audio. The reason is NAT and quirky residential routers. If you think about it, when your S3 is inside the firewall and connected to WiFi, it will have an IP address on your private LAN just like your Asterisk server. When your S3 is outside your firewall on either a cellular connection or someone else’s WiFi network, it will have an IP address that is not on your private LAN. Others may be smarter than we are, but we couldn’t figure a way to have connections work reliably in both scenarios using most residential routers. You can configure your S3’s PBX extension for NAT=No or NAT=yes, but you can’t tell Asterisk how to change it depending upon where you are. One simple solution is to deploy these phones with a VPN connection to your Asterisk server sitting behind a NAT-based firewall. The more reliable solution is to build your PBX in a Flash server in the cloud with no NAT-based firewall. Then use an IPtables WhiteList (aka Travelin’ Man 3) to protect your server. From there, you can either interconnect the cloud-based server with a second PBX behind your firewall, or you can dispense with the local PBX entirely. Either way will eliminate the NAT issues with missing audio. In both cases, use NAT=yes for the vMobile extension.

Another wrinkle involves text messaging. Traditional text messages work fine; however, MMS still is problematic unless you initiate the outbound MMS session with the other recipient. It’s probably worth noting that Google Voice never got MMS working at all despite years of promises. This wasn’t a deal breaker for us, but it’s a bug that still is being worked on.

Finally, there’s Sprint. You either love ’em or hate ’em. We really haven’t used Sprint service in about eight years. In the Charleston area, the barely 3G service still is just as lousy as it was eight years ago. But, if you live in an area with good Sprint coverage and performance, this shouldn’t be an issue for you. And vMobile works fine in Charleston. You just won’t be surfing the web very often unless you have hours to kill… waiting. Additionally, dialing numbers with less than 4 numbers is a non-starter with Sprint, but we’ll show you a simple workaround to reach 3-digit local extensions from your vMobile device below.

With a service as revolutionary as vMobile with Sprint’s new FMC architecture, we can’t help thinking there may be other cellular carriers with an interest in deploying this technology sooner rather than later. But, given the vMobile feature set, Sprint is good enough for now especially when WiFi connectivity is available almost everywhere.




vMobile Configuration at Vitelity

For the Vitelity side of the setup, you first configure your smartphone using the (included) My Phone app. When the application is run, your cellphone number will be shown. Tapping the display about a dozen times will cause the phone’s setup to be reconfigured. Vitelity will provide you the secret key to activate your account. Next, you’ll log into the Vitelity portal and choose vMobile -> My Devices under My Products and Services. The account for your vMobile device will already exist. Clicking on the pull-down menu beside your vMobile device will let you create your SIP account on Vitelity’s server. Enter the IP address or FQDN of your Asterisk server and set up a very secure password. Your username will be the 10-digit phone number assigned to your vMobile phone. Save your settings and then choose the Edit option to view your setup. The portal will display your Username, Password, and FreePBX/Asterisk Connect Host name. Write them down for use when you configure your new extension using FreePBX®.




vMobile Configuration for Asterisk and PBX in a Flash

On the PBX in a Flash server, use a browser to open FreePBX. Choose Applications -> Extensions and add a new generic SIP device. For Display Name and User Extension, enter the 10-digit phone number assigned to your vMobile device. Under Secret, enter the password you assigned in Vitelity’s vMobile portal. Click Submit and reload FreePBX when prompted. Then edit the extension you just created. Set NAT=yes and change the Host entry from dynamic to the FQDN entry that was shown in Vitelity’s vMobile portal, e.g. 7209876542.mobilet103.sipclient.org. Update your configuration and restart FreePBX once again. Finally, from the Linux command prompt, restart Asterisk: amportal restart. If you’re using a WhiteList with IPtables such as Travelin’ Man 3, be sure to add a new WhiteList entry for your vMobile Host entry. Finally, add your vMobile extension to any desired Inbound Routes to make certain your vMobile device rings when desired.

You now should be able to place and receive calls on your vMobile device. If you want to be able to call 3-digit Asterisk extensions on both WiFi and while roaming on the Sprint cellular network, then you’ll need to add a little dialplan code since Sprint reserves 3-digit numbers for emergency services and will reject other calls with numbers of less than 4 digits. Here’s the simple fix. Always dial 3-digit extensions with a leading 0, e.g. 0701 to reach extension 701. We’ll strip off the leading zero before routing the call. The dialplan code below works whether you’re calling a local 3-digit extension or a 3-digit extension on an interconnected remote Asterisk server. Simply edit extensions_custom.conf in /etc/asterisk and insert the following code at the top of the [from-internal-custom] context. Then restart Asterisk: amportal restart. Note that we’ve set this up so that, if you have an extension 701 on both the local server and a remote server, the call will be connected to the local 701 extension. If you have different extension prefixes for different branch offices (e.g. 7XX in Atlanta and 8XX in Dallas), then this dialplan code will route the calls properly assuming you’ve configured an outbound route with the appropriate dial pattern for each branch office.

exten => _0XXX,1,Answer
exten => _0XXX,n,Wait(1)
exten => _0XXX,n,Set(NUM2CALL=${CALLERID(dnid):1})
exten => _0XXX,n,Dial(sip/${NUM2CALL})
exten => _0XXX,n,Dial(local/${NUM2CALL}@from-internal)
exten => _0XXX,n,Hangup

Vitelity vMobile Special for Nerd Vittles Readers

Now for the icing on the cake… We asked Vitelity if they would consider offering special pricing to Nerd Vittles readers and PBX in a Flash users. We’re pleased to report that Vitelity agreed. By using this special link when you sign up, the vMobile monthly fee will be $8.99 instead of $9.95. In addition, your first month is free with no activation fee. We told you last week that there was a very good reason for choosing Vitelity as your SIP provider. Now you know why.

And, if you’re new to Cloud Computing, take advantage of the RentPBX special for Nerd Vittles readers. $15 a month gets you your very own PBX in a Flash server in the Cloud. Just use this coupon code: PIAF2012. Enjoy!

Originally published: Thursday, May 15, 2014




Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…

  1. RentPBX also is a corporate sponsor of the Nerd Vittles and PBX in a Flash projects. []

4 Months in Paradise: The Return of Free International VoIP Calling

With the impending implosion of Google Voice, it seemed appropriate to begin our quest for alternative termination providers. One of the real beauties of VoIP technology is you don’t have to put all of your eggs in one basket particularly in the termination department. It costs almost nothing to set up accounts with multiple providers for outbound calling. In addition to redundancy, the other clear advantage in using multiple providers for outbound calls is that you can take advantage of special rates to different destinations. So here’s the bargain of the week. If you have loved ones traveling to South America, Europe or Asia this summer, now’s your chance to sign up for VoIP service with FreeVoipDeal and enjoy four months of free calling to more than 50 countries around the world for every $15 of credits you purchase on their web site. Please note the fine print: "FreeVoipDeal reserves the right after a certain amount of calls to start charging the default rate." There is no mention of what that "certain amount" happens to be. When your free calling finally ends, you can either purchase $15 of additional credits for 120 more "free" days or continue to call all of the previously free destinations for about 2¢ a minute.

The company behind FreeVoIPDeal is betamax which hosts over 30 sites offering varying deals to different countries. BEWARE: The prices change regularly. So a country that’s free today may suddenly cost money tomorrow. How does a mere mortal keep track? Well, betamax probably hopes that you won’t. But an enterprising individual named Robert Siemer has done the work for you. His backsla.sh/betamax web site automatically updates the pricing for all betamax sites every day! If this sounds like a lot of work to save a few cents a minute, you’d be right. And Vitelity which sponsors both the Nerd Vittles and PBX in a Flash projects offers consistently low rates to all of these countries. You’ll find a DID special at the end of this article, and their excellent international rate table is available at this link.

Setting Up an Account. Before you can set up a trunk in PBX in a Flash, you’ll first need to create a FreeVoipDeal account. In the "old days" this required use of their Windows client to obtain your credentials. Now you can simply create an account on the web site at this link. You’ll need either a regular land line or a cell phone number to verify your registration. Once you’re set up and you’ve deposited at least 10 euros (about $15) in your account, it’s time to set up a SIP trunk and outbound route in PBX in a Flash.

Configuring a Trunk with PBX in a Flash. Assuming you already have a phone registered to an extension in PBX in a Flash, it’s a one-minute drill to configure a trunk and outbound route to support FreeVoipDeal. Using a browser, log into FreePBX® using your maint username and password. Choose Connectivity -> Trunks -> Add SIP Trunk. Name the trunk: FreeVoipDeal. For the Dialed Number Manipulation Rules, enter Prepend: 1 and Match Pattern: NXXNXXXXXX. Clear out all of the default entries in Outgoing and Incoming Settings. Then, in Outgoing Settings, enter Trunk Name: freevoipdeal. For the PEER Details, enter the following using your actual account USERNAME and PASSWORD. Then SAVE your settings and reload FreePBX.

username=USERNAME
authuser=USERNAME
secret=PASSWORD
type=peer
qualify=yes
nat=yes
insecure=port,invite
host=sip.freevoipdeal.com
fromdomain=sip.freevoipdeal.com
dtmfmode=auto
disallow=all
canreinvite=no
allow=ulaw

There’s no need to enter a CallerID number. All of the outgoing calls will be delivered as ANONYMOUS. You also won’t need to register with the provider since Asterisk® can handle this on the fly using your credentials entered above.

Configuring an Outbound Route with PBX in a Flash. One more step, and you’ll be ready to start making calls. Choose Connectivity -> Outbound Routes. For the Route Name, enter: FreeVoIPDeal. For the Dial Pattern to make U.S. calls, enter: NXXNXXXXXX. If you want to force callers to dial a prefix to use the FreeVoipDeal trunk, then enter a 9 or some other number in the Prefix field. For Trunk Sequence 0, choose: FreeVoipDeal. Click Submit Changes and restart FreePBX when prompted. You’re done!

Making Your First Call. Using a phone or softphone logged into your server, dial the prefix (if any) plus the 10-digit number of someone in the United States. When the called party answers, make sure you can hear the called party and vice versa. If not, open Settings -> SIP Settings in FreePBX and add your External IP and Local Network settings. Also make certain the NAT entry is set to YES.

Configuring Your Server for International Calls. We do not recommend configuring your server to permit international calls to everywhere. The reason is simple. If strangers manage to access one of your extensions, they can run up your phone bill in a hurry. For this reason, we also strongly recommend that you do not configure automatic credit card replenishment with any VoIP provider!

For international calling, we recommend you add a separate Dial Pattern to both your FreeVoipDeal trunk AND the outbound route for each country code you wish to enable. Here is the complete list of codes. For example, to allow calls to Germany from another country, you’d add 49XXXXXXXXXX, save your changes, and reload FreePBX.

Spoofing Your CallerID. If you first verify that you own a number by using the web portal, you then can spoof the outbound CallerID using the number you verified. Just add the following entries to your trunk settings replacing 9991234567 with your verified CallerID number. Special thanks to @hillclimber on the PIAF Forum for the tip.
fromuser=0019991234567
sendrpid=yes

Originally published: Friday, April 25, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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…