Tag Archives: piaf

Whether you’re just getting started with VoIP telephony or merely want to kick the tires of PBX in a Flash™ 2.0.6.3.1, this guide is for you. We’ll try to cover the basics as well as the fine points to get your PBX in a Flash system running on almost any platform. Let’s begin by telling you why PIAF™ is different and why it matters. PBX in a Flash is the only Asterisk® and FreePBX® aggregation in which most of the components are compiled as part of the install. PIAF is also the only platform that lets you choose a stand-alone server platform, or a turnkey virtual machine appliance for any Windows, Mac, Linux, or Solaris desktop, or a hosted platform from RentPBX for only $15 a month! If you’re cost conscious or just want to tinker, we’ve got a simplified version with dozens of utilities for the $35 Raspberry Pi.

Why does this matter? First, you get the very latest updates to the CentOS® 6.3 operating system. Second, you get your choice of numerous Asterisk versions including 1.8, 1.8-Certified, 10, and 11. Each is compiled from source on the fly. Third, you get your choice of FreePBX versions as well: 2.8, 2.9., 2.10, or 2.11. Fourth, you have a bloat-free platform that will let you easily add and compile almost any Linux or Asterisk add-on in a matter of minutes but only if you need it. One size fits all simply doesn’t work for everybody in the VoIP world. Fifth, you can adjust and fine-tune the existing PIAF setup to meet your own requirements any time you like. This was especially handy last week when Digium announced that all prior versions of Asterisk had some major security flaws. By tweaking a simple script, any PIAF server could be updated to the latest version of Asterisk in under 30 minutes without losing any of the existing configuration. Of course you’ll also have access to the largest collection of free Asterisk utilities and add-ons anywhere on the planet. If you don’t need a particular function, don’t add it. If you do, it’s there for the taking on the PIAF Forum and can be installed in minutes. For the newbies that just want a system that works, you can run the Incredible PBX script to generate a turnkey system that’s ready to plug in phones once you complete the installation. In less than 5 minutes, you’ll have over 50 Asterisk applications with free Google Voice calling in the U.S. and Canada. Last, but not least, we have the best VoIP resource and collection of experts you’ll find anywhere on the PIAF Forum. And it’s free!

Thinking back to when I began exploring VoIP telephony, I would suggest your first order of business should be to have a seat in a quiet room and carefully sketch out your objectives. Many want to test the waters privately before recommending VoIP telephony to an employer. If that’s your situation, then start with the Incredible PBX 11 Virtual Machine for VirtualBox. Once you’ve downloaded the software, you can install it and be up and running on almost any modern desktop computer in under 5 minutes. It also works with VMware. We have an airline using PBX in a Flash worldwide that started in just this way.

About half of our readers don’t live in the United States. They live in 201 other countries scattered around the globe (see our Flag Counter). But they’re unable to take advantage of Google’s free phone service offering for calls in the U.S. and Canada. That’s an integral part of all PIAF installs although you certainly can use any SIP or IAX provider as well as traditional PSTN lines if you want to. If that’s your situation, take a careful look at the RentPBX Cloud offering. They’re the best in the business and also one of our project sponsors. For $15 a month, you can enjoy everything that’s available to any U.S. resident with a hosted PBX in a Flash or Incredible PBX account on one of their servers. It’s a great way to take advantage of Google Voice and establish a corporate presence in the U.S.

For those that just want a cheap VoIP solution with lots of bells and whistles for your home or home office, a $35 Raspberry Pi running Incredible PBX can’t be beat. You get almost the full feature set of Asterisk for the cost of an evening meal. What’s included? We could write a book about it. In fact, we almost have. Start here. Then go here.

For the rest of you, the adventure begins here. We’re going to assume that you need a VoIP telephony solution that will support an office of up to several dozen employees and that you have an Internet connection that will support whatever your simultaneous call volume happens to be. This is above and beyond your normal Internet traffic. To keep it simple, you need 100Kbps of bandwidth in both directions for each call.¹ And you need a router/firewall that can prioritize VoIP traffic so that all your employees playing FarmVille won’t cause degradation in VoIP call quality. Almost any good home router can now provide this functionality. Remember to disable ALG on your router, and it’s smooth sailing.

For computer hardware, you’ll need a dedicated machine. There are many good choices. Unless you have a burning desire to preserve your ties with Ma Bell, we recommend limiting your Ma Bell lines to your main number. Most phone companies can provide a service called multi-channel forwarding that lets multiple inbound calls to your main number be routed to one or more VoIP DIDs much like companies do with 800-number calls. If this works for you, then any good dual-core Atom computer will suffice. You’ll find lots of suggestions in this thread. And the prices generally are in the $200-$400 range. For larger companies and Asterisk’s capacity with beefier hardware, see these stress test results.

If your requirements involve retention of dozens of Ma Bell lines and complex routing of calls to multiple offices, then we would strongly recommend you spend a couple thousand dollars with one of our consultants. They’re the best in the business, and they do this for a living. They can easily save you the cost of their services by guiding you through the hardware selection process. They also have turnkey phone systems using much the same technology as you’ll find in PBX in a Flash. You won’t hurt our feelings. 🙂

We get asked this question about a hundred times a week on the forums so here goes. There are more than two dozen permutations and combinations of CentOS, Asterisk, and FreePBX to choose from when you decide to deploy PBX in a Flash. We always recommend the latest version of CentOS because it tends to be the most stable and also supports the most new hardware. You have a choice to make between a 32-bit OS or 64-bit. Our preference is the 32-bit platform because it is better supported. The performance difference is virtually unnoticeable for most VoIP applications. With Asterisk, we always recommend an LTS release because those have long-term support. That narrows your choices to Asterisk 1.8 or the just released Asterisk 11. If you plan to use Digium® Phones (and we’ll get to that), then you’ll want either Certified Asterisk 1.8 or Asterisk 11. The conventional wisdom in the Asterisk community has been to avoid just released Asterisk versions like the plague. We think we’ve turned the corner on that approach. Asterisk 1.8 is close to end of life, and with Asterisk 11, you’re in great shape from a support standpoint for many years to come. We personally run Asterisk 11 and have yet to find something that functionally would qualify as a show stopper. That’s not to say there aren’t some bugs and security issues from time to time. A pretty serious collection of them was found just last week, but it affected all versions of Asterisk. So… our bottom line is that Asterisk 11 is the latest and greatest with the best feature set. If we were building a system for a commercial business, it would be our hands-down choice. In the PBX in a Flash world, we have colors for various versions of PBX in a Flash that support different versions of Asterisk. Asterisk 11 happens to be PIAF-Green, Asterisk 1.8=PIAF-Purple, Asterisk 10=PIAF-Red, Certified Asterisk 1.8=PIAF-Brown.

If there is one thing that will kill any new VoIP deployment, it’s choosing the wrong phones. If you value your career, you’ll let that be an organization-driven decision after carefully reviewing at least 6-12 phones that won’t cause you daily heartburn. You and your budget team can figure out the price points that work in your organization keeping in mind that not everyone needs the same type of telephone. Depending upon your staffing, the issue becomes how many different phone sets are you and your colleagues capable of supporting and maintaining on a long term basis.

On February 1, Schmooze Com will release a public beta of their commercial End Point Manager (EPM) at a price point of $25 per server. They’ve been using the application internally to support their commercial customers for over a year so it is not your typical beta software. Suffice it to say, it’s the best $25 you will ever spend. You can sign up for an account with Schmooze through our commercial support site and purchase the software as soon as it becomes available. After taking a look at the Admin User Guide, if you’re a true pioneer, drop us a note and we’ll get you a sneak peek. The beauty of this software is it gives you the flexibility to support over 150 different VoIP phones as well as other devices almost effortlessly. Using a browser, you can configure and reconfigure almost any phone on the market in a matter of minutes. So the question becomes which phones should you show your business associates. That again should be a decision by you and your management and budget teams, but collect some information from end-users first. Choose a half dozen representative users in your company and get each of them to fill out a questionnaire documenting their 10 most frequent daily phone calls and listing each step of how they processed those calls. That will give you a good idea about types and variety of phones you need to consider for different groups of users. Cheaper rarely is better. There’s a reason that everybody bought IBM Selectric typewriters. It had nothing to do with cost. And phones can last a very long time, even lousy ones. So don’t blow it!

The phone brands that we would seriously consider include Cisco, Aastra, Snom, Digium, Mitel, Polycom, Yealink, and Grandstream. Do you need BLF, call parking or multiple line buttons, a hold button, conferencing, speakerphone, HD voice, power over Ethernet support, distinctive ringtones for internal and various types of external calls, Bluetooth, WiFi, web, SMS, or email access, an extra network port for a computer, headset support, customizable buttons (how many?), quick dial keys, custom software, XML provisioning, VPN support? How easy is it to transfer a call? Do you need to mimic key telephones? Also consider color screens, touch screens, busy lamp indicators, extension modules (what capacity?). What do we personally use: several Digium phones of various types, a couple of Aastra phones, a Grandstream GXP2200, and a collection of Panasonic cordless DECT phones, a fax machine, and Samsung Galaxy Note II connected through an OBi202 with an OBiBT Bluetooth Adapter to our PIAF server. Good luck!

With the office politics out of the way, let’s get to the fun stuff.

• Download PIAF 2.0.6.3.1 ISO from SourceForge
• Burn the ISO to a USB Thumb Drive or a DVD using a Mac or Windows machine
• Boot dedicated server using PIAF Flash Drive or DVD

For most deployments, choose the default install by pressing Enter.

Leave the UTC System Clock option unchecked and pick your Time Zone. Tab to OK and press Enter.

Choose a very secure Root Password. Tab to OK and press Enter. Your server will whir away for 5-10 minutes installing CentOS 6.3. When the reboot begins, remove the DVD or USB thumb drive.

For today, we’re installing PBX in a Flash. So leave it highlighted, tab to OK, and press Enter.

Now pick your PIAF flavor, tab to OK, and press Enter.

The PIAF Configuration Wizard will load. Press Enter to begin.

Unlike any other aggregation, PIAF gives you the opportunity to fully configure Asterisk using make menuconfig if you know what you’re doing. For everyone else, type N and then confirm your choice.

Next, you’ll need to choose your Time Zone again for PHP and FreePBX. Don’t worry if yours is missing. A new timezone-setup utility is available in /root to reconfigure this to any worldwide time zone.

Next, choose your version of FreePBX to install. Ignore the screen info regarding Incredible PBX. It’s out of date. The following limitations apply if you plan to also install Incredible PBX and Incredible Fax:

Incredible PBX 3 requires PIAF-Purple and FreePBX 2.9
Incredible PBX 4 requires PIAF-Purple and FreePBX 2.10 (32-bit only)
Incredible PBX 11 will require PIAF-Green and FreePBX 2.11

Finally, you need to choose a very secure maint password for access to FreePBX using a browser. You can pick your own, or the installer will generate one for you. Don’t forget it.

The installer will give you one last chance to make changes. If everything looks correct, press the Enter key and go have lunch. Be sure you have a working Internet connection to your server before you leave. 😉

In about an hour, your server will reboot. You should be able to log in as root using your root password. Write down the IP address of your server from the status display (above) and verify that everything installed properly. Note that Samba is disabled by default. If you want to use your server with Windows Networking, run configure-samba once your server is up and running and you’ve logged in.

Most PIAF Configuration is accomplished using the FreePBX Web GUI. Point your browser to the IP address shown in the status display above to display your PIAF Home Page. Click on the Users tab. Click FreePBX Administration. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in the Config Module phase of the PBX in a Flash installation procedure above.

If you’re new to Asterisk and FreePBX, here’s the one paragraph primer on what needs to happen before you can make free calls with Google Voice. You’ll obviously need a free Google Voice account. This gets you a phone number for people to call you and a vehicle to place calls to plain old telephones throughout the U.S. and Canada at no cost. You’ll also need a softphone or SIP phone to actually place and receive calls. YATE makes a free softphone for PCs, Macs, and Linux machines so download your favorite and install it on your desktop. Phones connect to extensions in FreePBX to work with PBX in a Flash. Extensions talk to trunks (like Google Voice) to make and receive calls. FreePBX uses outbound routes to direct outgoing calls from extensions to trunks, and FreePBX uses inbound routes to route incoming calls from trunks to extensions to make your phones ring. In a nutshell, that’s how a PBX works. There are lots of bells and whistles that you can explore down the road.

To get a minimal system functioning to make and receive calls, here’s the 2-minute drill. You’ll need to set up at least one extension with voicemail, and we’ll configure a free Google Voice account for free calls in the U.S. and Canada. Next, we’ll set up inbound and outbound routes to manage incoming and outgoing calls. Finally, we’ll add a phone with your extension credentials.

A Few Words About Security. PBX in a Flash has been engineered to run on a server sitting safely behind a hardware-based firewall with NO port exposure from the Internet. Leave it that way! It’s your wallet and phone bill that are at stake. If you’re running PBX in a Flash in a hosted environment with no hardware-based firewall, then immediately read and heed our setup instructions for Securing Your VoIP in the Cloud Server. We would encourage you to visit your PIAF Home Page regularly. It’s our primary way of alerting you to security issues which arise. You’ll see them posted (with links) in the RSS Feed shown above. If you prefer, you can subscribe to the PIAF RSS Feed or follow us on Twitter. For late-breaking enhancements, you also should regularly visit the Bug Reporting & Fixes Topic on the PIAF Forum.

Extension Setup. Now let’s set up an extension to get you started. A good rule of thumb for systems with less than 50 extensions is to reserve the IP addresses from 192.x.x.201 to 192.x.x.250 for your phones. Then you can create extension numbers in FreePBX to match those IP addresses. This makes it easy to identify which phone on your system goes with which IP address and makes it easy for end-users to access the phone’s GUI to add bells and whistles. In FreePBX 2.10 or 2.11, to create extension 201 (don’t start with 200), click Applications, Extensions, Generic SIP Device, Submit. Then fill in the following blanks USING VERY SECURE PASSWORDS and leaving the defaults in the other fields for the time being.

Write down the passwords. You’ll need them to configure your SIP phone.

Extension Security. We cannot overstress the need to make your extension passwords secure. All the firewalls in the world won’t protect you from malicious phone calls on your nickel if you use your extension number or something like 1234 for your extension password if your SIP or IAX ports happen to be exposed to the Internet.

In addition to making up secure passwords, the latest versions of FreePBX also let you define the IP address or subnet that can access each of your extensions. Use it!!! Once the extensions are created, edit each one and modify the permit field to specify the actual IP address or subnet of each phone on your system. A specific IP address entry should look like this: 192.168.1.142/255.255.255.255. If most of your phones are on a private LAN, you may prefer to use a subnet entry in the permit field like this: 192.168.1.0/255.255.255.0 using your actual subnet.

Adding a Google Voice Trunk. There are lots of trunk providers, and one of the real beauties of having your own PBX is that you don’t have to put all of your eggs in the same basket… unlike the AT&T days. We would encourage you to take advantage of this flexibility. With most providers, you don’t pay anything except when you actually use their service so you have nothing to lose.

For today, we’re going to take advantage of Google’s current offer of free calling in the U.S. and Canada through the end of 2013. You also get a free phone number in your choice of area codes. PBX in a Flash now installs a Google Voice module under FreePBX -> Connectivity that lets you set up your Google Voice account with PBX in a Flash in just a few seconds once you have your credentials.

A Word to the Wise: All good things come to an end… especially those that are free. So plan ahead with some alternate providers that keep your phones working should Google decide to pull the plug or change the terms with Google Voice.

Signing Up for Google Voice. You’ll need a dedicated Google Voice account to support PBX in a Flash. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

We’ve tested this extensively using an existing Gmail account rather than creating a separate account. Take our word for it. Inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So… set up a dedicated Gmail and Google Voice account², and use it exclusively with PBX in a Flash. Google Voice no longer is by invitation only. If you’re in the U.S. or have a friend that is, head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for PBX in a Flash to function with Google Voice! Otherwise, inbound and/or outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings and enable it. Be sure to try one call each way from Google Chat in Gmail. Then disable Google Chat in GMail for this account. Otherwise, it won’t work with PIAF.

While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Configuring Google Voice Trunk in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize PBX in a Flash to meet your specific needs. Click the Connectivity tab in FreePBX 2.11 and choose Google Voice [Motif]. To Add a new Google Voice account, just fill out the form. NOTE: The form has changed from prior releases of FreePBX. Do NOT check the last box: Send Unanswered to GoogeVoice Voicemail, or you’ll have problems receiving incoming calls.

Google Voice Username is your Google Voice account name without @gmail.com. Password is your Google Voice password. NOTE: Don’t use 2-stage password protection in this Google Voice account! Phone Number is your 10-digit Google Voice number. Next, check only the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit Changes and reload FreePBX. Down the road, you can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill. For Google Apps support, see this post on the PIAF Forum.

Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. It also provides redundancy which costs you nothing if you don’t use the backup providers. The Google Voice module actually configures an Outbound Route for 10-digit Google Voice calling as part of the automatic setup. If this meets your requirements, then you can skip this step for today.

Inbound Routes. An Inbound Route tells PBX in a Flash how to route incoming calls. The idea here is that you can have multiple DIDs (phone numbers) that get routed to different extensions or ring groups or departments. For today, we’ll build a simple route that directs your Google Voice calls to extension 201. Choose Connectivity -> Inbound Routes, leave all of the settings at their default values except enter your 10-digit Google Voice number in the DID Number field. Enable CallerID lookups by choosing CallerID Superfecta in the CID Lookup Source pulldown. Then move to the Set Destination section and choose Extensions in the left pull-down and 201 in the extension pull-down. Now click Submit and save your changes. That will assure that incoming Google Voice calls are routed to extension 201.

IMPORTANT: Before Google Voice calling will actually work, you must restart Asterisk from the Linux command line interface. Log into your server as root and issue this command: amportal restart.

Eliminating Audio and DTMF Problems. You can avoid one-way audio on calls and touchtones that don’t work with these simple settings in FreePBX: Settings -> Asterisk SIP Settings. Just plug in your public IP address and your private IP subnet. Then set ULAW as the only Audio Codec.

General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. In FreePBX 2.11, choose Admin -> Module Admin and click on the Upgrade Notifications shield on the right. Plug in your email address, click Submit, and save your changes. Done!

Setting Up a Desktop Softphone. PBX in a Flash supports all kinds of telephones, but we’ll start with the easy (free) one today. You can move on to "real phones" once you’re smitten with the VoIP bug. For today, you’ll need to download a softphone to your desktop PC or Mac.

The easiest way to get started is to set up a YATE softphone on your Desktop computer. Versions are available at no cost for Macs, PCs, and Linux machines. Just download the appropriate one and install it from this link. Once installed, it’s a simple matter to plug in your extension credentials and start making calls. Run the application and choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of your server, 201 for your account name, and whatever password you created for the extension. Click OK.

Once you are registered to extension 201, close the Account window. Then click on YATE’s Telephony Tab and place your first call. It’s that easy!

Monitoring Call Progress with Asterisk. That about covers the basics. We’ll leave you with a tip on how to monitor what’s happening with your PBX. There are several good tools within the FreePBX GUI. You’ll find them under the Reports tab. In addition, Asterisk has its own Command Line Interface (CLI) that is accessible from the Linux command prompt. Just execute the following command while logged in as root: asterisk -rvvvvvvvvvv.

What’s Next? We’ve barely scratched the surface of what you can do with PBX in a Flash. Log into your server as root and type help-pbx for a list of simple install scripts that can add almost any function you can imagine. And Incredible PBX 11 is just around the corner. You can try it out in the Incredible PBX 11 Virtual Machine today.

Special Thanks. In just ten days, Nerd Vittles kicks off its 9th year of publication. We wanted to express our thanks to all of you for visiting. In just the last three months, we’ve again begun compiling a list of countries that are represented in the Nerd Vittles Fan Club. Nerd Vittles has spread to 204 awesome countries so far. Just click on the Google Map and have a look for yourself. Very inspiring. Thanks again for your support and Happy New Year!

Originally published: Thursday, January 10, 2013

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

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

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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.

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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. There obviously are ways to dramatically reduce VoIP bandwidth. Here’s a table that provides the details. The tradeoff is that you’ll need a beefier computer to compress and decompress VoIP packets if you elect to use a codec such as G.729. [↩]
2. You also can use a dedicated Google Apps account for Google Voice with the latest version of the FreePBX module. Don’t use your regular Google Apps email address with Google Voice, or inbound calling will not work! [↩]

We’ve just returned from AstriCon 2012 in Atlanta. What a show! Not only were the exhibits plentiful and eye-opening for the breadth of coverage that Asterisk® has garnered, but the dozens of sessions on virtually every facet of Asterisk were first-class. If you haven’t been, I would encourage you to clear your calendar for a few days next October. Mark Spencer was a proud papa as, indeed, he should be.

There’s been a good bit of turnover at Digium® in the past year, and it was great to finally put names with faces. The new blood is a welcome addition. To a person, they were professional, personable, and all about making Asterisk a better product. They fully appreciate that Asterisk’s future success hinges on broader community participation. David Duffett became the Community Director last June and wasted little time mending fences. Not only did he reach out to those of us that package Asterisk distributions (PBX in a Flash, Elastix, and FreePBX® Distro to name a few), but he also put together the first (ever) face-to-face meeting between the distribution players and the Asterisk developers. It was time well spent and provided all of us a better appreciation for the challenges in managing a project as complex as Asterisk. Matt Jordan, who now heads up the open source development group, has broad experience from the commercial sector, and it showed. It was crystal clear that the days of the core development train wrecks that blew everyone’s Asterisk dialplans out of the water were little more than a historical footnote.

While there is a core development team that extends beyond Digium, make no mistake. Where Digium goes, so goes Asterisk. When you peel away the management, marketing, and support layers at Digium, what struck me was how thin these folks really are spread. While there are upwards of 100 employees at Digium, the staff is fairly evenly split between two organizational units, Asterisk and Switchvox. The staff is also divided geographically with the Switchvox team still operating out of California while the Digium headquarters remains in Huntsville, Alabama. Our educated guess is about 10% of that staff is actually dedicated to software development. You may recall driving down the highway and seeing a road construction crew of 10 people where only one guy had a shovel. Digium clearly isn’t that model. Management, sales, and support really matter in the software and hardware development business. It’s the way every successful technology business operates… if you want to survive. Infrastructure matters! Insofar as Asterisk is concerned, it reinforces the critical necessity of focus and prioritizing objectives.

With that in mind, Digium has made what we believe was a wise decision in scrapping the Asterisk SCF project. You may recall this was the engineering effort to build a fully-redundant Asterisk platform so that, when one server failed, another took over without missing a beat. Out of a million Asterisk production servers, the question becomes how many sites really need this level of redundancy if a spare server is sitting in the rack. Suffice it to say, SCF was consuming enormous programming resources for a project with must-have appeal to an infinitesimally small segment of Asterisk’s installed base. The good news is that it frees more programmers to work directly on Asterisk 11 and 12 while bringing some of the SCF technology into the main Asterisk project. Think SIP!

And, speaking of Asterisk 11 and 12, Asterisk 11 is officially on the street. As with Asterisk 1.8, it’s another release with long-term support. The good news for us is that Jingle, Jabber, and Gtalk have been reworked into a new, integrated channel driver: chan_motif. And it works! Full support for Google Voice in FreePBX will be available very, very soon. In fact, Andrew Nagy had it humming along while we were at AstriCon.

More good news from AstriCon 2012 was the arrival of Digium’s Application Development Toolkit for Digium Phones. As promised, this new firmware lets developers build customized JavaScript applications to run on the Digium Phones themselves. Digium has even built several samples to get everyone started. We were lucky enough to snag one of the new phones during the random drawing so we’ll build you a couple of additional apps in coming weeks just for fun.

Last, but not least, on the hardware front… Raspberry Pi’s were everywhere. The recent addition of a 512MB Raspberry Pi at the same $35 price point is going to make this device a real contender in the SOHO Asterisk market. We actually had ours running with a Verizon MiFi device during the show. You could make calls, play an IVR, and get a weather report in the basement of a hotel in downtown Atlanta. Impressive! So get yours ordered and come join the party.

Finally, a word about the AstriCon audience. It looked to us to be a crowd of close to 1,000 people. We spoke to dozens and dozens of in-house developers both from large corporations as well as Asterisk support organizations that maintain thousands of Asterisk servers throughout the world. We were impressed by the scope and sheer magnitude of this untapped expertise with a treasure trove of Asterisk code. If it were somehow made available to the community, it instantly could propel Asterisk to the next plateau without Digium having to lift a finger. Now all we need to do is figure out how to harness that talent pool and their code for the benefit of everyone that depends upon Asterisk to meet their communications needs. Happy Halloween!

Originally published: Wednesday, October 31, 2012  

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

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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.
 

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