Posts tagged: Internet/Web

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.

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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. []

Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

It’s been an exciting couple of weeks watching the overwhelmingly positive response to our release of Orgasmatron 5.1. With this version, we introduced a new Asterisk® security model that took into account the ever-increasing security risks posed by exposing web and telephony servers to direct Internet access. The bottom line is this. If your telecom requirements still can be accomplished by placing a server securely behind a $35 hardware-based Internet firewall with no Internet exposure, then it makes absolutely no sense to dangle such a tempting target in front of the world’s most nefarious creeps.

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Our experience suggests that the only trade off with this new approach is the inability to receive anonymous SIP calls… a small price to pay considering the potential financial and computer risks involved. You still can place outbound VoIP calls as well as placing and receiving calls using any of the phone numbers registered on your new PBX in a Flash server. And, thanks to Google Voice, SIPgate, and IPkall, all inbound calls are free, and all outbound calls to numbers in the U.S. and Canada are free as well.

If a SIP URI and your own Freenum/ISN number are simply features you can’t live without, sign up for a voip.ms IAX account, and you’ll get a SIP URI for free. Inbound SIP URI and Freenum/ISN calls will set you back $1 for every 1,000 minutes billed in 6 second increments.

Or you can sign up for a free IP Freedom CallCentric account and configure a new SIP trunk in FreePBX by following these directions. Once configured, your new server SIP URI will be 1777xxxxxxx@in.callcentric.com where xxxxxxx is your assigned 7-digit CallCentric number.

Keep in mind that a new security vulnerability has been found with either Asterisk or FreePBX almost monthly. The chart below tells you why. With virtually limitless attack surfaces because of the number of interrelated components in CentOS, Asterisk, and FreePBX comes enormous and recurring potential for remote compromise of these systems. Rather than play this cat-and-mouse security game with the underworld, the Orgasmatron design changes the paradigm. It lets you use any (secure or insecure) version of Asterisk and FreePBX without worrying about any outside attacks. Do passwords on your new server matter? Not really… unless there is someone inside your firewall that you don’t trust. :roll: Are we going to secure them anyway? Absolutely. But instead of the constant worry over new security vulnerabilities, Orgasmatron 5.2 lets you enjoy exploring the world of Asterisk and VoIP telephony with an incredibly rich feature set that you won’t find anywhere else, period! We’ll resist making any other device analogies, but the idea here is to protect the good guy (you!) while keeping the bad guys out. No penetration. No worries. Simple as that.

In our former life working for a living, we actually procured and managed multimillion dollar PBXs as part of our “other duties as assigned.” Without qualification, we can tell you that the feature set that Orgasmatron 5.2 brings to the table for free runs circles around anything you could buy (then or now) in the commercial marketplace. And, at one time or another, we purchased every Nortel feature good money could buy. There’s one other difference. Orgasmatron 5.2 runs swimmingly on a $200 Atom-based PC that you can purchase at any Best Buy as well as hundreds of other stores including Amazon, NewEgg, and Buy.com. We paid more than $200 to provision an additional extension on our Nortel switch! You, of course, can add as many extensions as you like. De nada.

So, why a new version of Orgasmatron in only a few weeks? Well, it’s not security-related. In fact, there is nothing wrong with continuing on with Orgasmatron 5.1. Unfortunately, it relied exclusively upon SIPgate to make free Google Voice calls in the U.S. and Canada. And SIPgate required an invite using an SMS message from a U.S.-based cellphone. That pretty well knocked out all of our friends living outside the United States. Today’s version fixes that by letting anyone sign up for a free IPkall phone number in Washington state. All you need is a valid email address. The setup process is a bit more complex because IPkall doesn’t support registered connections to their servers. But we’ll walk you through the additional steps and, once completed, your server will be just as secure as the SIPgate approach we set up with Orgasmatron 5.1. And few, if any, Linux skills are required to set up or manage Orgasmatron 5.2. As we’ve noted previously, if you can handle slice and bake cookies, you’ve got the necessary skillset! Be aware this is about a one-hour project, and you need to track through the article carefully, or the entire house of cards comes down.

New Asterisk Security Model. Orgasmatron 5.2 maintains our design goal of running an absolutely secure Asterisk PBX from behind a hardware-based firewall with either NO INBOUND PORTS exposed to the Internet with SIPgate or an IP-address-restricted IAX port for IPkall. Don’t defeat this security mechanism by exposing additional ports on your PBX in a Flash server to Internet access. And choose your NAT-based firewall/router carefully. All of these devices are not created equally. Not only do some perform better than others, but certain models are notoriously bad at handling NAT-based routing tasks, a critical requirement in the Asterisk VoIP environment. In almost every case of problems with one-way audio, the real culprit can be traced back to a crappy router. For $35, you really can’t go wrong with the dLink WBR-2310. If you want traffic shaping functionality as well, take a look at dLink’s Gaming Router, our personal favorite.

As long as your router, Google Voice, SIPgate, and IPkall passwords are secure, you can sleep like a baby. We use an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada because Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you’re calling is connected. The SIPgate or IPkall trunk is used by Google Voice to call you back so the inbound call is always free. We handle the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just refrain from using extension 75 in Asterisk for personal conferencing!

The choice is yours. You can use SIPgate with no incoming ports exposed to your server from the Internet. Or you can use IPkall and map UDP port 4569 (IAX2) on your hardware-based firewall to the internal IP address of your new PBX in a Flash server. Even with the IPkall setup, we’ve locked down IPtables (our Linux firewall) to restrict IAX access to several specific IP addresses so your server remains absolutely secure. We’ve also included support for FonicaTec’s IAX offering for those that want a backup IAX provider. We’ll have much more to say about IPtables in coming weeks.

If you’ve already installed Orgasmatron 5.1 and it’s working for you, do you need to upgrade? NO. With the exception of the new IAX support for IPkall, the code in Orgasmatron 5.2 is identical.

We, of course, continue to recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.

Swiss Army Knife Inventory. There’s no need for a Swiss Army Knife if you don’t know what all the blades are for. So, for those that are wondering what’s included in the Orgasmatron 5.2 build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.

Prerequisites. Here’s what you’ll need to get started:

  • Broadband Internet connection
  • Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
  • $200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
  • Free Google Voice account (HINT: Under $2 on eBay)
  • Free SIPgateOne residential account (Use cell to get SMS invite) OR
  • Free IPkall IAX account

Learn First. Install Second. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today’s VoIP world. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.

Today’s Drill. The installation process is straight-forward, but a little different than the Orgasmo 5.1 scenario because of the need to accommodate IPkall. Just don’t skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:

1. Install the latest version of PBX in a Flash
2. Run the Orgasmatron 5.2 Installer
3. Configure a softphone or SIP telephone
4. Configure Providers for Orgasmatron 5.2
5. Enter your Google Voice and SIPgate/IPkall credentials
6. Change existing passwords to secure your system

Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta on a new Atom-based PC. This beta is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use the identical payload for versions 1.3, 1.4, 1.5, and 1.6 of PBX in a Flash. The beta label simply means we haven’t had time to sufficiently test CentOS. But this is not a Microsoft-style beta so fear not!

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot from the installation CD and type ksalt to begin.

WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can’t find the kickstart file. Just tab to OK and press Enter. Don’t change the name or location of the kickstart file! This will get you going. Think of it as a CentOS ‘feature’. :-)

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes

When prompted, change the ARI password to something really obscure. You’re never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source on the Orgasmatron builds because of parking lot issues in the latest releases of Asterisk.

Running the Orgasmatron 5.2 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.2 installer:

cd /root
wget http://pbxinaflash.net/orgasmo52.x
chmod +x orgasmo52.x
./orgasmo52.x

Have another 15-minute cup of coffee. It’s a great time to consider a modest donation to the Nerd Vittles project. You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Now run passwd-master1. Set your FreePBX passwords to something very secure but different from your Linux root password.

Next, type status2 and press Enter. Write down the IP address of your new server.

If you’re using IPkall, now’s the time to log in to your hardware-based firewall/router and map UDP port 45693 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don’t worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you’re using SIPgate instead of IPkall.

For good measure, we recommend you reboot your server at this point. The command to type is simple: reboot4

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone, and you’ll find lots of recommendations on Nerd Vittles. For today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 82812661 as the password for extension 701 and the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Don’t Forget! After you change your extension passwords later in this tutorial, you will need to update the password entry in X-Lite, or you will no longer be able to place calls! In fact, you will get locked out of your server for 90 minutes after three failed password attempts. So put this on a sticky note so you don’t forget, or you’ll regret it in about 15 minutes.

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. And there’s really no reason you can’t use both if they’re available in your location. Do NOT use special characters in your provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we’d recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You’ll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don’t worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn’t matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you’ll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You’ll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you’ve opted to use IPkall, here’s the drill. First, you’ll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we’ll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don’t have an available number, your request apparently goes in the bit bucket. You’ll know because IPkall typically turns these requests around in a few minutes. Don’t worry about the mothership entry. We’ll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it’s dynamic, then you’ll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you’ve got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you’ll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It’s worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

We’re going to be making some entries in FreePBX which is the web-GUI that manages PBX in a Flash. For now, we simply need to enter your new IPkall phone number so that incoming calls to your IPkall number will actually ring on your softphone. Later, we’ll make some further adjustments once we get Google Voice humming along.

Using a web browser from your desktop, log in to FreePBX 2.6 at the following link substituting your server’s private IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master).

When FreePBX loads, choose Setup, Trunks, ipkall (iax). In the USER Context field, enter your 10-digit IPkall phone number. Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it’s a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Now let’s test your new phone number. Call your IPkall number from a cellphone or some other phone. Your softphone should ring. Answer the call, and be sure you have voice in both directions! Do not proceed without success here, or the rest of the adventure is a waste of your time.

Configuring Google Voice. Google Voice still is by invitation only so the first thing you’ll need is an invite. If you’re in a hurry, then stroll over to eBay where you’ll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you’ve chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.

Google then will place a call to your number and ask you to enter a confirmation code that’s been provided. When your cellphone (SIPgate) or softphone (IPkall) rings, answer it and punch in the number. Wait for confirmation. Then hang up.

As we mentioned earlier, there’s no reason you can’t set up both SIPgate and IPkall forwarding numbers in Google Voice. Just repeat the drill with the other provider’s number if you wish to activate both numbers for use with Google Voice. They’re not both going to ring simultaneously as you will see in a minute.

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

  • Call ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF

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

Finally, place a test call to your new Google Voice number and be sure your cellphone or softphone rings. Don’t move forward until you’ve been able to successfully place a call to your phone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to PBX in a Flash. We’re ready to insert your Google Voice credentials and SIPgate/IPkall number into PBX in a Flash. You’ll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate or IPkall RingBack DID (beginning with a 1). Don’t get the 10-digit GV number mixed up with the 11-digit SIPgate/IPkall RingBack DID, or nothing will work. :-)

Log back into your server as root and issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

Configuring FreePBX. Now shift back to your Desktop and, using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master). Depending upon which intermediate provider you’re using, do the following:

SIPgate Setup. When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:

7004484f0:B8TTW3@sipgate.com/4155201234

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

SIPgate and IPkall Setup. While still in FreePBX with your browser, click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate or IPkall Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

Securing FreePBX. You’re almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.

Don’t forget to adjust your X-Lite password to match the password entry you made for extension 701!

Orgasmatron Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let’s place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then, in the subsequent articles below, we’ll cover in detail how to customize every application that’s been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we’d recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

The VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O – Nerd Vittles Orgasmatron Demo (running on your PBX)
  • 1234*1061 – Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 – Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P – Enter a five digit zip code for any U.S. weather report
  • 6-1-1 – Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 – Get the latest news and sports headlines from Yahoo News
  • T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
  • F-A-X – Send a fax to an email address of your choice
  • 4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L – Record a message and deliver it to any email address
  • C-O-N-F – Set up a MeetMe Conference on the fly
  • 1-2-3 – Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 – ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 – ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 – Schedule a hotel-style wakeup call from any extension
  • 1061*1061 – PBX in a Flash Support Conference Bridge
  • 882*1061VoIP Users Conference every Friday at Noon (EST)


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


Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you’ll want to activate CallerID Lookups in FreePBX. Choose Setup, CID Superfecta, Default and enter the maint password you created with passwd-master. Then choose Tools, Module Administration, CallerID Lookup, Enable, Process and Save the Settings. Then edit each of the Inbound Routes and choose CallerID Superfecta as the CID Lookup Source. Save your changes. Finally, choose Setup, CallerID Lookup Sources, CallerID Superfecta and be sure your maint password created with passwd-master is correct here, too. If not, update it. For additional tips, visit the forums.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you’ve heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It’s also perfect for off-site backups!

Also check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches for Orgasmatron 5.1 and 5.2. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.

Coming Attractions. In our next episode, we’ll walk you through the process of adding a second, third, fourth, and fifth Google Voice line to your server so that you’ll never run out of free calling on your server. Enjoy!




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


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


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


Some Recent Nerd Vittles Articles of Interest…

  1. passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. []
  2. status is the PIAF utility program that displays the current status of most major applications running on your server. []
  3. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device “inside” your firewall for further processing. []
  4. reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. []

VoIP Over VPN: Securely Interconnecting Asterisk Servers

We’ve just returned from a week in the Pacific Northwest teaching an Asterisk® course for an organization that wants to interconnect satellite offices using Asterisk servers. This coincided with a support request from one of America’s premier airlines which wants to do much the same thing for all of its reservation counters in airports situated in feeder cities around the country. Suffice it to say, PBX in a Flash in conjunction with Asterisk and Hamachi VPNs is perfectly suited to let anyone build these interconnected systems in minutes rather than months. In fact, with less than a day’s worth of introduction to Asterisk and PBX in a Flash, a group of 16 network administrators with no previous Asterisk experience did just that in a one-hour lab session during our training seminar last week. At the risk of (further) destroying our ability to earn a living, here’s how we did it.

Proxmox as a Training Tool. Before we get into the nitty gritty of actually interconnecting Asterisk servers with Hamachi VPNs, let us provide the free tip of the week for those of you that want to experiment with interconnecting Asterisk servers or for those that like to test various Asterisk scenarios without rebuilding servers all day long. There is no finer tool for this than the Proxmox Virtual Environment, a free and easy to use Open Source virtualization platform for running Virtual Appliances and Virtual Machines. With a sale-priced Dell T105 with a Quad Core AMD Opteron processor and 8 gigs of RAM, you’ll have a perfect platform to run about 16 simultaneous PBX in a Flash servers. The trick is finding the machines on sale for half price which is about every other week. Our lab system which matches this configuration was less than $600 with RAM purchased from a third party. You can save most of the shipping cost by using our coupon link in the right column to shop at Dell’s small business site.

Proxmox lets you build virtual machines in two ways: OpenVZ templates or Qemu/KVM Templates and ISO images. While we intend to offer an OpenVZ template for PBX in a Flash soon, currently it’s easy to create your own ISO template using the standard PBX in a Flash ISO image. Once you’ve uploaded your ISO image into Proxmox, simply create a new virtual machine by giving it a name, specifying 512MB of RAM and a 30GB partition. In 10 seconds or less, your new VM will be ready to boot. Start your VM and then open the VNC console window within the Proxmox web interface and install PBX in a Flash just as if you were building a stand-alone machine. When the 15-minute install completes, run through the Orgasmatron Installer setup, and you’ll have your turnkey PBX in a Flash system ready for production in less than 30 minutes.

You don’t have to repeat this drill for every virtual machine. Instead, use the built-in Proxmox backup utility to make a backup image of what you built. Shut down the VM, create a /backup directory, and then schedule the compressed backup in the web browser. When the backup completes, you’ll have a backup image in /backup with a file name like this: vzdump-101.tgz.

To create a new virtual machine, you issue the following command while positioned in the /backup directory specifying the number for the new virtual machine:

vzdump --restore vzdump-101.tgz 102

In about 3 minutes, you’ll have a second virtual machine that’s a clone of the first one. Because it’s a true clone, it would obviously have the same MAC address for the virtual NIC. You don’t want that or all of your VMs would boot up using the same IP address. Using the Proxmox web interface, just edit the new VM 102 by switching from the Status tab to the Hardware tab, delete the existing Ethernet device, and then create a new Ethernet device under the hardware address list pulldown. This will create a new virtual NIC with a new MAC address. So, when you boot VM 102, it will be assigned a new IP address by your DHCP server. You can decipher the new IP address by opening the VNC console window for VM 102 after you boot it up. Now you’re an expert. You can create the additional Baker’s Dozen turnkey PBX in a Flash servers in about an hour. Start all of them up, and you’ve got an instant training facility and PBX in a Flash playground.

April, 2012 Update. See our new article for a current state-of-the-art VoIP VPN.

Creating Hamachi VPN. You obviously don’t need a virtual private network in order to interconnect Asterisk servers. But, as easy as the Hamachi VPN is to set up, especially with PBX in a Flash servers, why wouldn’t you want all of your inter-Asterisk communications secured and encrypted? In addition to the capacity limitation of the Proxmox server, there’s another reason we chose to build 16 PBX in a Flash VMs. That happens to be the number of servers you can interconnect with the Hamachi Virtual Private Network without incurring a charge.1 Why use the Hamachi VPN when OpenVPN is free with unlimited network connections and no strings? The short answer is it’s incredibly simple to set up without public and private key hassles, and it supports dynamic IP server addressing with zero configuration. We plan to cover OpenVPN in a subsequent article but, for many implementations, Hamachi VPNs offer a robust, flexible alternative that can be deployed in minutes.

If you’re not using PBX in a Flash, there are a million good Hamachi VPN tutorials available through a quick Google search. If you are using PBX in a Flash, we’ve done the work for you. With the Orgasmatron Installer build, you’ll find the Hamachi VPN installation script in /root/nv. For other PBX in a Flash systems, just download the install-hamachi.x script from here or, after logging into your server as root, issue the following commands:

wget http://pbxinaflash.net/source/hamachi/install-hamachi.x
chmod +x install-hamachi.x
./install-hamachi.x

Before beginning the Hamachi VPN install, it’s a good idea to make yourself a cheat sheet for the servers you plan to interconnect. We’re going to interconnect 3 servers today, but doing 16 is just more of the same. You’ll need a unique name for your virtual private network. Pick a name that distinguishes this VPN from others you may build down the road. For our example, we’re going to use piaf-vpn. Next, you need a very secure password for your VPN. We’re going to use password for demonstration purposes only. Finally, you need a unique nickname for each of your servers, e.g. piaf-server1, piaf-server2, and piaf-server3 for our example setup today.

For the first Hamachi install, we’ll need to create the new network. For the remaining installs, we’ll simply join the existing network. Keep in mind that you can only remove machines from the network using the same server that was used to create the other VPN accounts initially so build out your virtual private network by starting with your main server, piaf-server1 in our example.

To begin the Hamachi VPN install, run the script using the commands shown above. Type Y to agree to the installer license and then press the Enter key to kick off the install. For the piaf-server1 install, type N to create a new Hamachi network. For the remaining installs, you’d type J to join an existing Hamachi network. Enter the network name you chose above. For our sample, we used piaf-vpn. Type it twice when prompted. Now type your network password and then your nickname for this server when prompted to do so. Then standby while the Hamachi software is installed. It takes a few minutes depending upon the speed of your network connection. And remember, do NOT use our sample network name. Make up your own and don’t forget it. When the install completes, you can review the log if you’d like. Unless something has come unglued, Hamachi should now be running on your first server. Repeat the drill on your other servers.

The next step is to grab some of our scripts to make it easier to manage Hamachi on your servers.

cd /usr/local/bin
wget http://pbxinaflash.net/source/hamachi/hampiaf
wget http://pbxinaflash.net/source/hamachi/hamachi-servers
chmod +x ham*
cd /root
wget http://pbxinaflash.net/source/hamachi/hamachi.faq

The hamachi.faq document provides all of the commands you’ll need to manage Hamachi including the steps to start over with a totally new virtual private network. For now, let’s be sure your network is running. Type: hamachi-servers piaf-vpn using the network name you assigned to your own VPN. Then type it again, and it should display all of the servers on your VPN with their private VPN IP addresses:

root@pbx:~ $ hamachi-servers piaf-vpn
This server:
Identity 5.151.123.1
Nickname piaf-server1
AutoLogin yes
OnlineNet piaf-vpn

Going online in piaf-vpn .. failed, already online
Retrieving peers’ nicknames ..
* [piaf-vpn]
5.151.123.2 piaf-server2
5.151.123.3 piaf-server3

Finally, a word of caution about security. One of the drawbacks of the ease with which you can create Hamachi VPNs is the ease with which you can create Hamachi VPNs. Anyone that knows your network name and password can join your network with one simple command. You can kick them off from the main server where the VPN was created (hampiaf evict piaf-vpn 5.249.146.66), but you can’t keep them from joining. So, protect your network by making the password extremely secure. There currently is no way to change your network password. All you can do is create a new network with a new network name and a more secure password.

Interconnecting Asterisk Servers. Once your VPN is established and all of your servers are on line, then we’re ready to interconnect them with Asterisk and FreePBX. There are a number of ways to do this. For smaller networks, we’re going to show you the easy and secure way using IAX and the VPN you just created. As with the VPN setup, a cheat sheet comes in handy to avoid erroneous entries that would cause your calls between servers to fail. What we recommend is assigning and creating a block of extensions on each of your servers with different ranges of numbers. For example, we’re going to use four-digit extensions in the 1xxx range for piaf-server1, 2xxx for piaf-server2, and 3xxx for piaf-server3. The idea here is that the extensions are unique between your servers. This makes it easy to dial between offices without having to resort to dialing prefixes. So the first step in interconnecting your servers is to build the necessary extensions on each of your servers.

Now for the cheat sheet. Using the hamachi-servers tool above, decipher the VPN IP address of each of your servers and make a chart with the server names, the range of extension numbers, and the VPN IP address of each server. You’ll also need to think up a very secure password. We’re going to use the same one for all of the servers although you certainly don’t need to. So long as the password you choose is secure, there’s really no reason not to use the same one.

piaf-server1 1xxx 5.151.123.1 password
piaf-server2 2xxx 5.151.123.2 password
piaf-server3 3xxx 5.151.123.3 password

Creating Trunks. The next step is to create an IAX trunk on each server for each remaining server in your network. In our example, on piaf-server1, we’d want to create trunks for piaf-server2 and piaf-server3. On piaf-server2, we’d want to create trunks for piaf-server1 and piaf-server3. And so on.

NOTE: Because of a change in IAX design to fix a security issue that arose after this article was originally published, be sure to add the following line in the User Details of each trunk below:

requirecalltoken=no


On your first server (piaf-server1 in our example), using a web browser, open FreePBX and choose Admin, Setup, Trunks and then click Add IAX2 Trunk. Create the trunk to piaf-server2 with the following entries. Leave everything blank except the entries shown below:

While still on piaf-server1, repeat the process to create a trunk for piaf-server3:

On your second server (piaf-server2 in our example), using a web browser, open FreePBX and choose Admin, Setup, Trunks and then click Add IAX2 Trunk. Create the trunk to piaf-server1 with the following entries. Leave everything blank except the entries shown below:

While still on piaf-server2, repeat the process to create a trunk for piaf-server3:

On your third server (piaf-server3 in our example), using a web browser, open FreePBX and choose Admin, Setup, Trunks and then click Add IAX2 Trunk. Create the trunk to piaf-server1 with the following entries. Leave everything blank except the entries shown below:

While still on piaf-server3, repeat the process to create a trunk for piaf-server2:

Creating Outbound Routes. Now we need to tell Asterisk how to route the calls between the servers. In a nutshell, we want calls to extensions in the 1xxx range routed to extensions on piaf-server1, calls to 2xxx extensions routed to piaf-server2, and calls to 3xxx extensions routed to piaf-server3. On each server, create an outbound route for each of the remaining servers. Name the routes server1, server2, and server3 as appropriate. The critical pieces of information in each outbound route are the dial string (which should match the extensions on the server we’re connecting to) and the Trunk Sequence (which should be the appropriate IAX trunk for the server we’re connecting to).

On piaf-server1, we’d have a server2 outbound route with a Dial String of 2xxx and a Trunk Sequence of IAX2/piaf-server2. Then we’d have another server3 route with a Dial String of 3xxx and a Trunk Sequence of IAX2/piaf-server3. If you have a catch-all outbound route, be sure to move these routes above the catch-all in the right column. Then reload your dialplan.

On piaf-server2, we’d have a server1 outbound route with a Dial String of 1xxx and a Trunk Sequence of IAX2/piaf-server1. Then we’d have another server3 route with a Dial String of 3xxx and a Trunk Sequence of IAX2/piaf-server3. If you have a catch-all outbound route, be sure to move these routes above the catch-all in the right column. Then reload your dialplan.

On piaf-server3, we’d have a server1 outbound route with a Dial String of 1xxx and a Trunk Sequence of IAX2/piaf-server1. Then we’d have another server2 route with a Dial String of 2xxx and a Trunk Sequence of IAX2/piaf-server2. If you have a catch-all outbound route, be sure to move these routes above the catch-all in the right column. Then reload your dialplan.

If you’re setting this up with PRI or T1 connections between your servers, you might also want to specify at least secondary trunk sequences for each of the outbound routes to provide some redundancy. For example, on piaf-server1, you might want a secondary Trunk Sequence for server2 that specified IAX2/piaf-server3. Then, if the primary connection between server1 and server2 was down, Asterisk would attempt to complete calls to 2xxx extensions by routing them to server3 and then on to server2 from there. To the caller and call recipient, they’d never know that the direct link between server1 and server2 had failed.

Alternate routing might also be appropriate where you have more capacity between certain servers. For example, if you had a single T1 line between server1 and server3 but you had PRI connections between server1 and server2 and between server2 and server3, then it might make more sense to indirectly route 3xxx calls from server1 through server2 and then on to server3 rather than the direct route from server1 to server3. Enjoy!



Free DIDs While They Last. Sipgate is giving away a free U.S. DID with free incoming calls plus 200 free minutes for outbound calls. Better hurry. Here’s the trunk setup for FreePBX-based systems:

Trunk name: sipgate

type=peer
username=ACCTNO
fromuser=ACCTNO
secret=ACCTPW
context=from-trunk
host=sipgate.com
fromdomain=sipgate.com
insecure=very
caninvite=no
canreinvite=no
nat=no
disallow=all
allow=ulaw&alaw

Registration Strong: ACCTNO:ACCTPW@sipgate.com/YOUR-DID-NUMBER

ACCTNO is the account number assigned to your sipgate account. ACCTPW is the password for your account. YOUR-DID-NUMBER is your 10-digit DID.

Finally create an inbound route using your actual 10-digit DID and assign a destination for the inbound calls.



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


Twitter Magic. If you haven’t noticed the right margin of Nerd Vittles lately, we’ve added a new link to our Twitter feed. If you explore a little, you’ll discover that the user interface now brings you instant access to every Twitter feed from the convenience of the Nerd Vittles desktop. Enjoy!


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


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


Some Recent Nerd Vittles Articles of Interest…

  1. See comment #1 below. []

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