Happy Halloween: Google Puts the Final Nail in the Google Voice Coffin


Meet the new Microsoft: It’s Google! If you follow Nerd Vittles regularly, you’ll recall that five months ago we warned you to start packing your bags if you were a Google Voice user with PBX in a Flash. And, as a special Halloween treat, Google gave the open source community the finger. As of May 15, 2014, there will be no more XMPP support for Google Voice. So much for embracing the open source community. And, for all of you that supported and fostered the Google Voice project and were part of Google’s development community, F U. Welcome to Oligopoly World!

It’s one thing for a company to shift gears and venture out in a new direction. It’s quite another to knife your developers in the back on the way out the door. Here’s an interesting background piece on the folks now driving the Google Voice Bus. Fundamentally, there’s so much wrong with Mr. Singhal’s posting that it’s difficult to know where to begin. But let’s start with “These [Google Voice] apps violate our terms of service.”

So let’s go to law school for a bit, shall we? One of the first things you’ll learn is never take the factual basis for someone else’s conclusions and insults at face value. Go read the material for yourself. And, when you do, what you’ll discover is there’s not even a hint that these Google Voice applications have ever violated Google’s terms of service. Never mind that Google was and still is selling and profiting from the GrooveIP application which is marketed and sold in the Google Play Store.

We’re reproducing the Terms of Service in all their glory so that you don’t have to take our word for it either. Of course, the terms are sprinkled across multiple web pages so bear with us…

And finally, there’s the Google catch-all which you will note changes regularly (and is about to change again). So, even if you didn’t violate the Terms last week, next week may be a different story. Multiply that Catch-22 by five years in the case of Google Voice.


Google Terms of Service

This Terms of Service will be replaced by our new Google Terms of Service effective November 11, 2013. Please see our summary of changes for additional details.
Last modified: March 1, 2012 (view archived versions)

Welcome to Google!

Thanks for using our products and services (“Services”). The Services are provided by Google Inc. (“Google”), located at 1600 Amphitheatre Parkway, Mountain View, CA 94043, United States.

By using our Services, you are agreeing to these terms. Please read them carefully.

Our Services are very diverse, so sometimes additional terms or product requirements (including age requirements) may apply. Additional terms will be available with the relevant Services, and those additional terms become part of your agreement with us if you use those Services.

Using our Services

You must follow any policies made available to you within the Services.

Don’t misuse our Services. For example, don’t interfere with our Services or try to access them using a method other than the interface and the instructions that we provide. You may use our Services only as permitted by law, including applicable export and re-export control laws and regulations. We may suspend or stop providing our Services to you if you do not comply with our terms or policies or if we are investigating suspected misconduct.

Using our Services does not give you ownership of any intellectual property rights in our Services or the content you access. You may not use content from our Services unless you obtain permission from its owner or are otherwise permitted by law. These terms do not grant you the right to use any branding or logos used in our Services. Don’t remove, obscure, or alter any legal notices displayed in or along with our Services.

Our Services display some content that is not Google’s. This content is the sole responsibility of the entity that makes it available. We may review content to determine whether it is illegal or violates our policies, and we may remove or refuse to display content that we reasonably believe violates our policies or the law. But that does not necessarily mean that we review content, so please don’t assume that we do.

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You may need a Google Account in order to use some of our Services. You may create your own Google Account, or your Google Account may be assigned to you by an administrator, such as your employer or educational institution. If you are using a Google Account assigned to you by an administrator, different or additional terms may apply and your administrator may be able to access or disable your account.

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Google’s privacy policies explain how we treat your personal data and protect your privacy when you use our Services. By using our Services, you agree that Google can use such data in accordance with our privacy policies.

We respond to notices of alleged copyright infringement and terminate accounts of repeat infringers according to the process set out in the U.S. Digital Millennium Copyright Act.

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When you upload or otherwise submit content to our Services, you give Google (and those we work with) a worldwide license to use, host, store, reproduce, modify, create derivative works (such as those resulting from translations, adaptations or other changes we make so that your content works better with our Services), communicate, publish, publicly perform, publicly display and distribute such content. The rights you grant in this license are for the limited purpose of operating, promoting, and improving our Services, and to develop new ones. This license continues even if you stop using our Services (for example, for a business listing you have added to Google Maps). Some Services may offer you ways to access and remove content that has been provided to that Service. Also, in some of our Services, there are terms or settings that narrow the scope of our use of the content submitted in those Services. Make sure you have the necessary rights to grant us this license for any content that you submit to our Services.

You can find more information about how Google uses and stores content in the privacy policy or additional terms for particular Services. If you submit feedback or suggestions about our Services, we may use your feedback or suggestions without obligation to you.

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When a Service requires or includes downloadable software, this software may update automatically on your device once a new version or feature is available. Some Services may let you adjust your automatic update settings.

Google gives you a personal, worldwide, royalty-free, non-assignable and non-exclusive license to use the software provided to you by Google as part of the Services. This license is for the sole purpose of enabling you to use and enjoy the benefit of the Services as provided by Google, in the manner permitted by these terms. You may not copy, modify, distribute, sell, or lease any part of our Services or included software, nor may you reverse engineer or attempt to extract the source code of that software, unless laws prohibit those restrictions or you have our written permission.

Open source software is important to us. Some software used in our Services may be offered under an open source license that we will make available to you. There may be provisions in the open source license that expressly override some of these terms.

The other important thing you learn in law school is that CONTEXT MATTERS. For example, there’s this generic warning buried in the terms which undoubtedly is the language Google is now using to claim that Asterisk, FreeSwitch, OBiHai, and GrooveIP implementations of Google Voice (among others) crossed the line:

Well, not so fast. It turns out that the “method” used for the Google Voice interface in all of these implementations happens to be the one that was documented, supported, and hosted on Google’s own developer web site. Here’s a snapshot of the site before it, too, disappears. Pay particular attention to the sentence we’ve highlighted: “[The Google Talk XMPP Extensions] are documented so that you can design a client that can take advantage of specific Google Talk features.”

It’s also worth recalling that Google Voice began as Craig Walker’s Grand Central project and that Craig, while heading up the Google Voice team, had the following dialogue with me in June of 2009:

WM: Just hoping that you might reconsider providing a SIP interface to Google Voice. It really would make it the Whole Enchilada. Adding a password through the web interface would make it very secure. We’ve written several articles on Nerd Vittles about Google Voice, and it would be an incredible addition.

CW: Hey Ward. Thanks for your note and I share a lot of your sentiments. Let me get it out to the world as my first priority:)

Here’s the Bottom Line: You can’t have it both ways, Google. You can’t encourage open source development with one hand and then use your other fist to slap down developers who do exactly what you encouraged them to do. Sorry to say it but this time Microsoft got it just about right:


For organizations considering a move to Google’s platform, you might want to think twice and take an open-minded look at the carnage in the Google Graveyard. Remember Google Buzz? How about Google Wave? Then came Google knol. How about Google Directory Assistance (GOOG-411)? And then Google Reader followed by iGoogle. And now the open source edition of Google Voice. This is just the tip of The Iceberg. And most of the open source Google apps have also fallen by the wayside: sports scores, weather reports, news feeds, and the JavaScript Maps API is on the chopping block for later this month. Many of these weren’t failed undertakings. Google simply decided they no longer wished to support them regardless of whether millions of people depended upon them. Ask yourself this simple question. Do you really want to stake the future of your company or your development efforts on this sort of corporate infrastructure? What can you do? Start here.

Where Do We Go From Here? Have no fear. The VoIP community is not going to wither away because of Google’s shenanigans. We’ll have a lot of great new approaches and solutions in coming weeks and months. In the meantime, make yourself a list of how you currently use Google Voice services. You’ll need that as we move forward. A belated Happy Halloween! It took me 5 days to cool off sufficiently to even write this article.

Originally published: Tuesday, November 5, 2013




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


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


Some Recent Nerd Vittles Articles of Interest…

Adventures in Twitterland: You Can’t Make This Stuff Up

Today is Twitter’s Coming Out Party with what could be 2013′s Biggest IPO. By the closing bell today, the U.S. probably will have a couple new billionaires. As one commenter on USA Today’s site noted, “I can’t get into the IPO. I’m too busy tweeting.” Unfortunately, we’re not.

Having just returned from a quick visit to New York earlier this week, we were greeted by the following message when we logged into our Twitter account:

Why is my account suspended? Well, it could be that you’ve violated “The Twitter Rules” or maybe Twitter suspects that your account was hacked or compromised. I’m reminded of the old playground adage: It’s For Me To Know and You to Find Out. Urban Dictionary has an interesting take on what that really means.

Imagine if you will that you’re walking down the street in your favorite town minding your own business when a cop grabs you by the collar and says, “You’re under arrest.” The natural inclination would be to ask why. Rather than tell you, the police officer hands you a copy of the municipal code with the admonition that you can always appeal.

Welcome to The World According to Twitter. Here you are assumed to know what you did wrong. And, if you don’t like the consequences, you can file “an appeal” after reading “The Rules”. All of that, of course, is perfectly fine if you actually broke “The Rules” and know what you did. Joseph Heller couldn’t have made this up. If you’ve read Catch 22, you’ll recall that “the term was introduced by Doc Daneeka, an army psychiatrist who invokes ‘Catch 22′ to explain why any pilot requesting mental evaluation for insanity — hoping to be found not sane enough to fly and thereby escape dangerous missions [during World War II] — demonstrates his own sanity in making the request and thus cannot be declared insane.”

We, of course, use Twitter for fun and to support our open source projects so there’s no financial hardship from Twitter’s antics while we endure “the appellate process.” But, put yourself in the position of a business person with thousands of followers and suppose you actually used Twitter to announce new products and merchandise sales. Don’t!

As you can see from the screenshot above, all of a sudden you have 0 Followers. So your entire business model is basically down the toilet at the sole discretion of Twitter. In case you’re curious, no, we haven’t violated any of Twitter’s rules. We still used to send a couple of tweets a day as we have since opening our account years ago. Has our account been hacked? Obviously not. We still can get into it, and there are no tweets from us offering you a million bucks for managing some African prince’s business affairs.

In becoming a public company today, one would hope this situation will be addressed. But then I’m reminded of our Comcast adventures and some of the horror stories you hear about Facebook, and it’s probably safe to conclude that the Borg was probably right: “Resistance is futile.” We’ve been patiently waiting 2 days for “our appeal” to be processed. We’ll let you know if we ever hear anything further. In the meantime, I’d hold off on that Twitter stock purchase.

If you happen to have a (working) Twitter account, do us and your friends a favor. Send a tweet to your followers about this article by clicking on the link below. Happy Tweeting!

1 p.m. Update. Surprise! Well, we’re back… kinda, sorta. Twitter let us out of jail but kept our pants and car keys. Following = 0, Followers = 0. No call. No email. Not even a tweet. Love means never having to say you’re sorry. Nice!

2:30 p.m. Update. Pants and car keys returned. Thanks a billion, Twitter. You’re the best.

Originally published: Thursday, November 7, 2013

The Definitive VoIP Quick Start Guide: Introducing PBX in a Flash 2.0.6.4.5

What a difference a year makes in the VoIP World! We now have a rock-solid, reliable Asterisk® 11 release and an equally stable FreePBX 2.11 on which to build state-of-the-art VoIP servers. If you’re new to the VoIP community, watch this video before you proceed.


Now let us welcome you to the World of PBX in a Flash™. This is our best release ever whether you’re a total newbie or an experienced Asterisk developer. You can’t really appreciate what goes into an open source product like PBX in a Flash until you try doing it yourself. The sad part is we and the CentOS™ development team are part of a dwindling few non-commercial entities that still are in the open source “business.” If you want to actually learn about Asterisk from the ground up using pure source code to customize your VoIP deployment, PBX in a Flash has no competition because your only other option is to roll your own starting with a Linux DVD. So our extra special kudos go to Tom King, who once again has produced a real masterpiece in that it is very simple for a first-time user to deploy and, at the same time, incredibly flexible for the most experienced Asterisk developer. The new PIAF 2.0.6.4.5 ISOs not only provide a choice of Asterisk® and FreePBX® versions to get you started. But now you can build and deploy standalone servers for SugarCRM™, NeoRouter™ VPN, YATE, and OpenFire™ XMPP using the 32-bit and 64-bit PIAF™ ISOs. So let’s get started.

Making a Hardware Selection

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 Angry Birds 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 to increase 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. :-)

Choosing the Right PIAF Platform

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 these have long-term support. That narrows your choices to Asterisk 1.8 or Asterisk 11. At this juncture, we think you’d be crazy to install anything other than Asterisk 11. It’s incredibly reliable and stable, and it will be supported for years to come. It also supports Digium Phones. The 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 11.6 happens to be the latest PIAF-Green, and we recommend you install it with the latest version of FreePBX as well, 2.11.0.11

Choosing the Right Phones

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.

Schmooze Com has released their commercial End Point Manager (EPM) at a price point of $39 per server. They’ve been using the application internally to support their commercial customers for over a year. Suffice it to say, it’s the best money you will ever spend. You can sign up for an account with Schmooze through our commercial support site and purchase the software now. You can review the Admin User Guide here. The beauty of this software is it gives you the flexibility to support literally hundreds of different VoIP phones and devices almost effortlessly. Using a browser, you can configure and reconfigure almost any VoIP phone or device 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. Keep in mind that phones can last a very long time, even lousy ones. So choose carefully.

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: Yealink’s T46G is our favorite, and we also have 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.

Installing PBX in a Flash

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

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 6.4. When the reboot begins, remove the DVD or USB thumb drive.

Log into your server as root from either the console or an SSH connection to the IP address displayed on your server. Unless you need to install custom hardware drivers, choose the first option to install PBX in a Flash.

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. You’ll note there’s a new color. :-)

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 also to reconfigure this to any worldwide time zone once the install has completed.

Next, choose your version of FreePBX to install. 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 requires PIAF-Green and FreePBX 2.11 (recommended!)

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. :wink:

In about 30-60 minutes, your server will reboot. You should be able to log in as root again 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.

Configuring PBX in a Flash

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. FreePBX now has some of the best documentation in the business. Start here.

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.

User Extension … 201
Display Name … Home
Outbound CID … [your 10-digit phone number if you have one; otherwise, leave blank]
Emergency CID … [your 10-digit phone number for 911 ID if you have one; otherwise, leave blank]

Device Options
secret … 1299864Xyz [randomly generated]
dtmfmode … rfc2833
Voicemail Status … Enabled
voicemail password … 14332 [make this unique AND secure!]
email address … yourname@yourdomain.com [if you want voicemail messages emailed to you]
pager email address … yourname@yourdomain.com [if you want to be paged when voicemail messages arrive]
email attachment … yes [if you want the voicemail message included in email]
play CID … yes [if you want the CallerID played when you retrieve message]
play envelope … yes [if you want date/time of the message played before the message]
delete Vmail … yes [if you want the voicemail message deleted after it's emailed to you]
vm options … callback=from-internal [to enable automatic callbacks by pressing 3,2 after playing a voicemail message]
vm context … default

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:

  • Call ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF
  • Call Options (Enable Recording)OFF
  • Global Spam FilteringON

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 may 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 and Incredible Fax can be installed in under 2 minutes to provide you almost every Asterisk application on the planet. You can read the complete tutorial here. In addition, Travelin’ Man 3 can be installed as part of Incredible PBX for rock-solid Internet security. If you care about your wallet, add Travelin’ Man to your server!

New App of the Week. We’re pleased to introduce Trunk Failure Email Alerts for Asterisk supporting SIP, IAX2, and Google Motif trunks. Just insert your email address in this little script and run it every hour as a cron job. You’ll get an email alert whenever any of your VoIP trunks fail. Enjoy!

VoIP Experts on Twitter. GetVoip.com has just released their list of The Top 50 VoIP Experts to Follow on Twitter. It’s a great read… but we may be biased. :wink:

Originally published: Monday, October 28, 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.


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


Some Recent Nerd Vittles Articles of Interest…

The 5-Minute PBX: PIAF-Green Virtual Machine for Windows, Mac, or Linux

In our never-ending trek to build the Perfect PBX™, we have a sneak peek for you today of the soon-to-be-released PBX in a Flash™ 2.0.6.4.5 featuring CentOS® 6.4 LAMP stack (32-bit), Asterisk® 11.5.1, and FreePBX® 2.11.0.11. The 2.0.6.4.5 release also has a number of new security patches including a new Linux kernel that’s been patched to eliminate the reported zero-day vulnerability. Once you download today’s appliance, you can have a turnkey PBX running under VirtualBox® on almost any desktop computer in less than 5 minutes. We’re not talking about a crippled telephony platform with limited functionality. What you’ll have is the same platform that hundreds of thousands of organizations use to run their corporate phone systems. And, if you want the Incredible PBX™ feature set with literally dozens of open source telephony applications including news, weather, stocks, tide reports, SMS messaging, free faxing with Incredible Fax™, telephone reminders, wakeup calls, and more then just add a couple minutes to run two one-click installers. Welcome to the world of open source!

The real beauty of PBX in a Flash has not been that someone with sufficient expertise couldn’t assemble something just as good or even better. Watch some of the AstriCon presentations if you have any doubts. The beauty of PIAF is it puts this technology down where the goats can get it. It provides a toolset that encourages further development by simplifying the learning curve for a broad cross-section of the VoIP community while not compromising functionality or flexibility. The source code for the major components is included in the build so you can customize and recompile Asterisk or load a new version of Asterisk or any additional Linux app in minutes without losing your existing setup.

If Voice Over IP technology is Greek to you, here’s a new 60-minute video tutorial that will tell you everything you need to know about this exciting, new technology before you begin the actual installation process:


As many of you know, we have literally hundreds of gurus on the PIAF Forum. That doesn’t mean any particular person or group knows everything. It’s merely a designation that a particular individual is an expert at something. The collective wisdom of the group is what makes PBX in a Flash as a project better because we’ve put in place a platform that experts from many different disciplines can build upon without needing to learn everything about everything. Simply stated, you can be a terrific chef without knowing how to build a stove!

Turning to Asterisk® 11 and FreePBX® 2.11, these releases are a remarkable step forward both in terms of toolset and in the amazing stability of the platform. For our part, we want to get our latest release of PBX in a Flash with CentOS 6.4, Asterisk 11.5.1 and FreePBX 2.11 release into as many hands as possible with a near zero investment in hardware and setup time.

The Ultimate VoIP Appliance: PIAF Virtual Machine for VirtualBox

Today brings us to a new plateau in the virtual machine development era. Thanks to the masterful work of Tom King on PBX in a Flash 2.0.6.4.5, we’re pleased to introduce a new product that can be installed in under 5 minutes and will run on any Windows PC, Mac, or Linux machine as well as Solaris. And, unlike the dedicated machine platforms and OpenVZ compromises of years past, today’s PIAF-Green Virtual Machine is state-of-the-art giving you everything a bare metal install from source code would have provided. Most importantly, the components are truly portable. They can be copied to a 4GB flash drive1 for the price of a good hamburger and installed from there onto any type of machine that happens to be in front of you. Five minutes later, you have a fully functional Asterisk server with FreePBX and exactly the same feature set and source code that you would have had doing a bare metal PIAF install to a dedicated server. And we’ve built this 32-bit production-ready PIAF-Green Virtual Machine with Asterisk 11.5.1 and FreePBX 2.11. No Internet access required to perform the install. Sound too good to be true? Keep reading or, better yet, try the PIAF appliance for yourself. The install process is simple:

  1. Download and install VirtualBox onto a Desktop Machine of your choice
  2. Download and double-click on the PIAF-Green Virtual Machine to import it into VirtualBox
  3. Select the PIAF-Green Virtual Machine in VirtualBox Manager Window and click the Start button

Introducing Oracle VM VirtualBox

We’re late to the party, but Virtual Box®, Oracle’s virtual machine platform inherited from Sun, is really something. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. As mentioned, our recommendation is to put all of the 100MB installers on a 4GB thumb drive. Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.

Installing the PIAF Virtual Machine

Step #1 is to download the PIAF-Green Open Virtualization Appliance (.ova) from SourceForge.

Step #2: Verify the checksums for the 32-bit .ova appliance to be sure everything got downloaded properly. To check the MD5/SHA1 checksums in Windows, download and run Microsoft’s File Checksum Integrity Verifier.

For Mac or Linux desktops, open a Terminal window, change to the directory in which you downloaded the .ova file of your choice, and type the following commands:

md5 PIAF-Green-32.ova (use md5sum for Linux)
openssl sha1 PIAF-Green-32.ova

The correct MD5 checksum for PIAF-Green-32.ova is 7691127afd065412e40429cee49a4738. The correct SHA1 checksum for PIAF-Green-32 is 9b3828649dc9644d046ef83cb227aea4c1473c65.

Step #3: Double-click on the downloaded .ova file which will begin the import process into VirtualBox. It only takes a couple minutes, and you only do it once. IMPORTANT: Be sure to check the Reinitialize the Mac address of all network cards box before clicking the Import button.

Once the import is finished, you’ll see a new PIAF virtual machine in the VM List of your VirtualBox Manager Window. You’ll need to make a couple of one-time adjustments to the PIAF-Green Virtual Machine configuration to account for differences in sound and network cards on different host machines.

Click on the PIAF-Green Virtual Machine in the VM List. Then click Settings -> Audio and check the Enable Audio option and choose your sound card. Save your setup by clicking the OK button. Next click Settings -> Network. For Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is ever necessary for your PIAF-Green Virtual Machine. The rest is automagic.

Running the PIAF Virtual Machine in VirtualBox

Once you’ve imported and configured the PIAF Virtual Machine, you’re ready to go. Highlight PIAF Virtual Machine in the VM List on the VirtualBox Manager Window and click the Start button. The PIAF boot procedure with CentOS 6.4 will begin just as if you had installed PBX in a Flash on a standalone machine. You’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your PIAF VM.

Here’s what you need to know. To work in the PIAF Virtual Machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. Always shut down PIAF gracefully! Click in the VM window with your mouse, log in as root, and type: shutdown -h now.

Run the PIAF Virtual Machine behind a hardware-based firewall with no Internet port exposure!

To begin, position your mouse over the VM window and left-click. Once the PIAF VM has booted, log in as root with password as the password. Change your root password immediately by typing passwd at the command prompt. Now set up a secure maint password for FreePBX as well. Type passwd-master. If you’re not in the Eastern U.S. time zone, then you’ll want to adjust your timezone setting so that reminders and other time-sensitive events happen at the correct time. While logged into your server as root, issue this command:

/root/timezone-setup

Next, use a browser to log into your PIAF server by pointing to the IP address of the PIAF VM that’s displayed in the status window of the CLI. Click on the User button to display the Admin choices in the main PIAF Menu. Click on the FreePBX option to load the FreePBX GUI. You will be prompted for an Apache username and password. For the username, use maint. For the password, use whatever password you set up with passwd-master.

Now read the latest PIAF Quick Start Guide and begin your VoIP adventure. Then you’ll want to do some reading on VirtualBox. We’ve barely scratched the surface. Setting up Headless VMs that run in the background on any server is a breeze. From the command line, here’s an article to get you started. But you also can start Headless VMs from within the GUI by highlighting the VM and clicking Shift->Start. Always shut down VMs gracefully: Close->ACPI Shutdown. You’ll find more great tips at virtualbox.org and GitHub.

One of the real beauties of VirtualBox is you don’t have to use a GUI at all. The entire process can be driven from the command line. Other than on a Mac, here is the procedure to import, configure, and run the PIAF-Green-32 Virtual Machine:
 
VBoxManage import PIAF-Purple.ova
VBoxManage modifyvm "PIAF-Green-32" --nic1 nat
VBoxManage modifyvm "PIAF-Green-32" --acpi on --nic1 bridged
VBoxHeadless --startvm "PIAF-Green-32" &
# Wait 1 minute for PIAF-Green to load. Then decipher IP address like this:
VBoxManage guestproperty get "PIAF-Green-32" /VirtualBox/GuestInfo/Net/0/V4/IP
# Now you can use SSH to login to PIAF-Green at the displayed IP address
# Shutdown the PIAF-Green Virtual Machine with the following command:
VBoxManage controlvm "PIAF-Green-32" acpipowerbutton

On a Mac, everything works the same way except for deciphering the IP address. Download our findip script for that.

Adding Incredible PBX 11 and Incredible Fax

You can read all about the Incredible PBX 11 and Incredible Fax feature set in our recent Nerd Vittles article. If you decide you’d like to add one or both to your PIAF-Green Virtual Machine, just log into your server as root and issue the following commands. NOTE: You must install Incredible Fax after installing Incredible PBX, or you will lose the ability to install Incredible PBX at a later time. With Incredible Fax, there are a number of prompts during the install. With the exception of the prompt asking for your local area code, just press Enter at every other prompt.

cd /root
wget http://incrediblepbx.com/incrediblepbx11.gz
gunzip incrediblepbx11.gz
chmod +x incrediblepbx11
./incrediblepbx11
./incrediblefax11.sh

The Incredible PBX 11 Inventory. For those that have never heard of The Incredible PBX, here’s the current 11.0 feature set in addition to the base install of PBX in a Flash with the CentOS 6.4, Asterisk 11, FreePBX 2.11, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Incredible Fax, NeoRouter and PPTP VPNs, and all sorts of backup solutions are still just one command away and may be installed using the scripts included with Incredible PBX 11 and PBX in a Flash. Type help-pbx and browse /root for dozens of one-click install scripts.

Originally published: Tuesday, October 22, 2013




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


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


Some Recent Nerd Vittles Articles of Interest…

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

It’s An Oligopoly, Stupid: What’s Wrong with Comcast Business Class Internet?

Let’s begin with what sounds like a fairy tale but turns out to be a nightmare. After watching your country invest hundreds of millions of dollars in taxpayer-subsidized infrastructure, you’ve finally decided it’s time to buy your own car. You visit the only car dealer in town and are told that all vehicles are leased, not sold, for a period of three years. Cars come in three models. Would you like a 200, 300, or 400 horsepower engine? You opt for the 400 horsepower model and, just as your new car sputters off the lot, you discover a 14-page list of Terms and Conditions in your glove box. The document reveals that the manufacturer doesn’t make any guarantees regarding the performance or reliability of your new vehicle. And, if you attempt to return the car in a couple months because of the vehicle’s unreliability or lousy performance, you agree to forfeit 75% of the entire cost of the 3-year lease. And, no, you cannot sublease or even give your crappy purple Scion1 to somebody else. Aside from the fact that Scion actually makes great automobiles with excellent warranties, the only real difference in this scenario and The World According to Comcast is the fact that, with a car, the item being leased becomes less valuable every day. With Comcast, prices continue to go up, and up, and up…


So perhaps you think the cellphone oligopoly is similar. The Bell Sisters could only wish. With a cellphone plan, the carriers actually subsidize the cost of your discounted cellphone by spreading the cost over a period of two years. Thus, their early termination fees which typically run $200 to $300 are closely tied to recovery of the subsidized cost of your discounted phone. With Comcast, the company is not providing any hardware that you don’t actually pay for either up front or on a pay-as-you-go basis. Build out costs are payable in advance. Cable modems are leased by the month. When you discontinue service, the cable modem is returned and handed out to the next poor sucker customer waiting in line.

Download (PDF, 95KB)

Early Termination Fees. So let’s calculate the fee that Comcast could impose if you decide after a couple months that your business can no longer survive on their “Business Class” level of service and performance. On the Business Internet D50 plan (note that there’s no mention in the contract that this has been touted by the sales rep as a 50Mbit down, 10Mbit up Internet service), the “discounted” cost with one static IP address is $125 per month for 36 months = $4,500. You used the service for two months which reduces the lease balance to $4,250. The 75% Early Termination Fee for the service you never used and for which Comcast made no representation as to performance or reliability works out to a whopping $3,187.50. Makes your $125 monthly cellphone bill sound like a bargain, doesn’t it?

According to Craig Moffet, an analyst at the Wall Street firm Bernstein Research, Comcast and Time Warner are making a 97 percent margin on their “almost comically profitable” Internet services. So this is clearly not a case of recovering infrastructure costs. After all, most of those were either paid or subsidized by federal, state, and local governments. This is simply an oligopolist doing what they do best in unregulated local markets with almost zero competition by regulatory design. It’s good old-fashioned price gouging! What a coincidence that Comcast also happens to be one of the “top ten” political contributors in the United States.

Internet Performance. The other glaring problem lies with Comcast selling tiers of service at different price points while providing no assurance that the performance levels will ever be met. We all appreciate that Internet performance can vary; however, the Comcast terms go far beyond that. If Comcast provided a 2400 baud modem level of performance for three years, our reading of the contract terms suggests that Comcast is fully within its rights even though the service was sold as offering 50 megabit download speeds. Comcast’s terms and conditions specifically disclaim any responsibility for achieving any performance measurement ever. In short, the speed designations allow Comcast to charge higher rates without offering anything of contractual value to the customer in return.

How’s the Service? Let us briefly replay the last 8 days of dealing with Comcast Business Class in our office. This all transpired while a Comcast sales rep was pitching a new 3-year contract as the only way for us to decouple our existing Business Class Internet “service” from our residential cable TV bill. This would allow us to once again get business class support without a 30-minute residential support run-around on every Business Class Internet support call, a highly touted (and necessary!) feature that actually worked during the first two years of our first contract.

Sunday, Oct. 6, 6 a.m. – Preparing to leave town for AstriCon 10. Internet dead.
Sunday, Oct. 6, 7 a.m. – Reset cable modem, Comcast tests modem. All fine. Internet still dead.
Monday, Oct. 7, all day – Repeat of Sunday. Internet still dead.
Tuesday, Oct. 8, all day – Same story.
Wednesday, Oct. 9, all day – Same story.
Thursday, Oct. 10, 4 p.m. – Another hour with Comcast support. Will try to schedule visit for Friday.
Friday, Oct. 11, 10 a.m. – Tech arrives. Takes one look at modem and declares the unit defective.
Friday, Oct. 11, rest of day – Internet works.
Saturday, Oct. 12, 6 a.m. – Internet dead. Comcast reports A-OK. Is the modem in bridge mode? Yes.
Sunday, Oct. 13, 3 p.m. – Comcast support: In bridge mode? Ooops. No. Internet finally works.
Monday, Oct. 14, 4 p.m. – Internet dead. Looks like a fiber cut. Offers 1 month Internet credit.
Monday, Oct. 14, 9 p.m. – Internet works.

What Can You Do About It? For openers, raise hell with your favorite Congressman. Assuming he or she didn’t receive a “political contribution,” it might actually help. Then write or visit your state and local elected representatives and hand them a copy of this article if you’re too shy to tell your own story. Nearly everybody has a ‘Comcast Story’ to tell. Encourage all of these folks either to open up the marketplace for real competition or to establish local initiatives to bring affordable Internet service to local businesses and communities. Last, but not least, write your local newspaper and encourage them to shine a spotlight on business practices such as these. You might be surprised by the results. If there’s an organization that deserves a lower job approval rating than Congress, we have a tip for you.

What’s Next? We’ve reached out to @ComcastCares for comment. We’ll let you know if there’s a meaningful response.

Originally published: Tuesday, October 15, 2013




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


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


Some Recent Nerd Vittles Articles of Interest…

  1. With apologies to Scion that actually makes perfectly functional and dependable automobiles! Listen to the song for details. []

AstriCon 10: WOW! What a Coming Out Party for Asterisk 12!

It was the tenth anniversary of AstriCon in Atlanta this week with an SRO crowd, and what a week it was. Comparing Asterisk® 12 to Asterisk 11 and previous iterations would be much like comparing Windows 8 to Windows 3.1. Facelift doesn’t begin to describe the metamorphosis. There’s a brand new (robust) SIP implementation featuring PJSIP, and a new restful interface known as ARI that lets you get at all of the Asterisk internals with a simple web command. You can transfer calls, play sound files, set up conferences and move participants with the click of a button. Here’s the complete slideshow on Asterisk 12. That’s just the tip of iceberg. Features still are being added almost weekly. Unlike previous releases, there’s no feature freeze in Asterisk 12 so long as the additions are germane to the new functionality already incorporated into Asterisk 12. See the Wiki link for details. And the work under the hood really shines. Here’s another shocker. There’s actually documentation. In fact, you can no longer add code to the project without also providing documentation for your code. Yes, this is truly a WOW moment! Think of it more as the Second Coming of Asterisk. Most importantly, the alpha release was relatively stable! And now there’s a beta.

We were so excited by our preliminary look at Asterisk 12-alpha and the new PIAF-Black (we’re running out of colors) preview that we actually set up our Incredible PBX IVR with news, weather, stocks, reminders, and conferencing in our car for the 5-hour trip to Atlanta. Speech to text works. Text to speech works. In fact, with the exception of CDRs, everything worked with the existing FreePBX® 2.11 dialplan code. The CDR transformation will get sorted out in coming weeks/months with the introduction of FreePBX 2.12. Back to our story, with the help of Verizon Wireless 4G LTE tethering, close to a hundred people called into the demo IVR to try things out for themselves. No crashes, no stuttering, high quality VoIP calls. In short, just pure Allison IVR goodness. See if you can find her in the snapshot of your favorite hams.

Monday and Tuesday of AstriCon are primarily devoted to future development goals and objectives. Known as AstriDevCon, the forum is by invitation only to about 50 developers. We were fortunate to be included this year, but the work of the group is no secret. You can review exactly what was said and where we’re headed in this Wiki posting of the meeting. The good news is there will be more meat on the bones of Asterisk 12. And our security recommendations regarding incorporation of Fail2Ban-like functionality directly into Asterisk made the list. This means attacks and vulnerabilities can be identified in real time rather than delaying a response until Fail2Ban has had time to scan your logs for nefarious activity. We also were heartened to see almost a third of AstriCon presentations devoted to beefing up security on Asterisk systems. We’ll have more to say about those discussions as well as some recommendations in coming weeks.


WebRTC also was a big hit at AstriCon 10. This provides the functionality to make and receive phone calls to Asterisk-based systems using nothing more than a web browser. A WebRTC implementation for FreePBX-based systems including PBX in a Flash is just around the corner. Some of you may recall that we released a WebRTC appliance based upon PBX in a Flash last winter. It will run on any Windows, Mac, or Linux desktop so read the tutorial and download away for a quick preview of what’s coming.

We were especially pleased to review Steve Murphy’s new SayScript proposal for Asterisk 13. In a nutshell, this would transform all “say” commands with prerecorded sound files into pluggable modules with translation to any language on the fly. “Language packs” would be developed that include script files, prompts, and SayScript logic to handle almost any speech chore in Asterisk. Think of it as Text-to-Speech on Steroids. Here’s a complete copy of Steve’s presentation in PDF format if you’d like more details.

Last but not least, Schmooze introduced a new fault-tolerant, high availability commercial module for FreePBX that brings complete redundancy to the Asterisk platform for the first time. Until October 15, you can purchase the two-node system for half-price, just $1,500. Be sure to let them know your friends at Nerd Vittles referred you so that we can keep the lights on for another year. And keep us posted during your deployment.

Originally published: Friday, October 11, 2013




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