In what can only be described as a telephony game changer, Google Voice this past weekend expanded the scope of its offering by providing transparent SIP connectivity through Gizmo5 for inbound and outbound calling. Simply stated, you now can connect virtually any telephone to Google Voice using a garden-variety Internet connection. And the phone can be almost any SIP telephone or a standard home telephone plugged into a $40 ATA. Letting folks make click-to-dial calls through a PC is too geeky for most. But today's offering is a new animal. Google Voice now works with regular telephones.

Did we mention that you get a free phone number of your choice in almost any area code? Did we mention that every call you make throughout the United States and Canada is free? And, believe it or not, transparent Asterisk® support works out of the box as well. If your bread and butter business is SIP termination services in the United States (Are you listening, Vonage and Comcast?), then today probably isn't going to be your lucky day. For everyone else, it may just be remembered as the most important telephony development since the breakup of Ma Bell's monopoly. And now it's clear why Google Voice reserved a million DIDs. They're going to need every one of them... and more! Meet your New Phone Company®, Goliath Google, Inc. What Google Voice was missing was a simple interface to standard telephones, softphones, and SIP. Gizmo5 provides all of those missing pieces... and so much more. How about an almost-free Skype interface for openers.

As many of you know, we were ecstatic when Google Voice arrived with free U.S. calling, voice mail transcription, and SIP connectivity to Asterisk. Our solution lasted less than a week until Google slammed the SIP door and spoiled our party. So we shifted gears and showed you how to use a free Gizmo account and a free Google Voice account to make free SIP calls using Asterisk. Well, that lasted about a week as well although Craig Walker, who founded GrandCentral and now serves as the Google Voice Product Manager, responded to my inquiry about SIP support saying it sounded like a good idea and they would consider it once the initial Google Voice rollout was complete. Guess what? They've kept their promise.

Ironically, we had planned to introduce a new Google Voice solution for Asterisk today and were putting the finishing touches on the article when this news broke over the weekend. We've decided to postpone that discussion because, frankly, the Google Voice-Gizmo5 SIP marriage is the right way to go. It's straight-forward. It's proven technology. It's rock-solid reliable. And it's FREE!

Newly discovered issues with both security and Gizmo5's business model as pertains to making calls through Google Voice have given us pause in recommending the solution described below. In a nutshell, the solution below requires that you provide your Google email credentials to Gizmo5 in order to make the connection to Google Voice for free unlimited 20-minute 3-minute calling. Late yesterday, Gizmo5 announced a new 2¢ per minute fee for Google Voice calling (now described as Gizmo Voice). Yuck!

Even if you don't mind a stranger having unfettered access to your Gmail account, your Google credentials also may be used for other Google services including Google Checkout. Without a clearly defined business relationship between Google and Gizmo5, this would be a huge security risk. Having read several articles which hinted at a business relationship between Google and Gizmo5, we put our security concerns aside. However, when Gizmo5 began changing the ground rules for these calls (almost daily), it raised red flags that Google might not, in fact, be either a business partner or even a willing participant in Gizmo5's creation. As events continued to unfold, we have discovered that Gizmo5 may, in fact, be using a connection process that is not unlike the one we had planned to introduce this week anyway. And we have no business relationship with Google.

Bottom Line: Whether you are using an Asterisk server or not, WAIT! We have an equivalent, secure solution which is now available at no cost. We recommend you disable your Gizmo5-Google Voice setup if you already have put it in place and change your Gmail password! Then read the new Nerd Vittles article for a secure way to connect to Google Voice for free calling.

Our plan today is to show you the easy way to connect Asterisk to Google Voice through Gizmo5 to make free outbound phone calls and to receive free incoming calls. We'll leave the setup for a SIP phone, a generic Asterisk server, and an analog adapter such as the PAP2T-NA for another day. But we'll get to them sooner rather than later.

So, altogether now, welcome back... Googlified Messaging™. Before we begin...

Accounting 101. We hear you asking, "How long can the calls be free?" The short answer is probably not forever but long enough to run just about everyone else out of the business. Beyond that, what we see in our crystal ball pretty much lines up with Tim O'Reilly's talk at OSCON last week. And, at some point, Google may give you a choice of paying for the calls or perhaps volunteering to be their guinea pig for the mother of all indexing experiments. You'd agree to let them record your voice calls without identifying you individually. Then they could transcribe and index all of the keywords in your conversation and use those to identify buying trends, favorite movies, whatever. Remember, you can already say "Pizza" on your iPhone and get a list of nearby pizza parlors so this isn't as far-fetched as you may think. And keep in mind that, in some states, you only need the permission of one party to a telephone conversation to make a recording. Thanks to Amazon, it's been quite a resurgence for Big Brother. We thought we'd join the party with a little Orwellian hypothesizing of our own.

Step #1. If you're starting from scratch, the easiest way to get everything working today including Asterisk is to begin by installing PBX in a Flash, and then run the Orgasmatron Installer. This puts all the pieces in the proper places, and you'll be up and running in under an hour. For the complete soup-to-nuts tutorial, start here.

Step #2. You obviously still need a free Google Voice account to use Google Voice or Google Voice Dialing through Gizmo5. So that's next. If you don't have a Google Voice account, you can request an invite here. Our non-scientific survey suggests that it's taking less than a month to get an invite after you apply. YMMV! Once you have a Google Voice account and a local phone number (Google has reserved a million of them so... not to worry!), then you're all set.

Step #3. Next, you need a Gizmo5 account. If you don't have one, you can sign up for one within FreePBX once you run the Orgasmatron Installer. Or, you can download a Gizmo5 softphone and sign up that way. We're not sure it's required, but be charitable. Put a little money in your Gizmo5 Call Out account. You'll have it for a rainy day or international calling.

Step #4. We'll set up at least one forwarding phone number in your Google Voice account to match your Gizmo5 number. You don't have to actually use it, but it does have to be registered as one of your GV forwarding numbers. Unlike our previous SIP tutorials about Google Voice, you no longer have to configure your Google Voice account to forward all incoming calls to voicemail. As you may recall, this allowed you to call your Google Voice number and press a few keys to make an outbound call instead of listening to your voicemails. With the new Google Voice-Gizmo5 SIP offering, you no longer have to jump through all those hoops. It's a straight SIP-to-SIP-to-SIP connection from your Asterisk server to Gizmo5 to Google Voice.

Step #5. To use Asterisk for incoming calls through Google Voice, you can designate a forwarding number in Google Voice that connects to one or more extensions on your Asterisk system whenever anyone calls your Google Voice number. All you really need for this is one DID. This could be your Gizmo5 number, or it could be a free IPkall or SIPgate DID that's pointed to an extension or ring group on your Asterisk server. Since all of these calls are free, the area code of the DID really doesn't matter. The only number that will really matter to your callers is your main Google Voice number so be sure to select one for your hometown. Incidentally, you can add other forwarding numbers in Google Voice that will ring simultaneously with the DID on your Asterisk server. This could be your vacation home, your cell phone, or even your office phone.

Getting Started. We're going to be jumping back and forth between your Google Voice account, your Gizmo5 account, and the FreePBX web interface to your Asterisk server. So open each account in a separate tab with your web browser. To keep things simple, we're going to assume that you'll be using your Gizmo5 account to connect to your Asterisk server. In Asterisk lingo, the Gizmo5 account looks like any other DID on your Asterisk system.

FreePBX Setup for Gizmo5. If you've run the Orgasmatron Installer, you'll have a new Gizmo5 Integration option under the Setup tab. When you click on that option, you have the choice of either creating a new Gizmo5 account or using your existing account. Fill in the blanks to activate or create your new Gizmo5 account.

Once you've logged in, click Gizmo5 Integration Main Page. Choose Send all calls (except local extensions) through Gizmo5 and click Update Outbound Routes. For the time being, make certain that you have a default inbound route that rings one or more functioning extensions on your Asterisk system. You have to be able to answer an incoming call to complete the next steps. Finally, click on the Outbound Routes option. In the far right column, move the Gizmo5 entry to the top of the list and reload your dialplan when prompted.

If you're using a FreePBX-based system that doesn't have the Gizmo5 Integration option, you'll first need to establish an account at Gizmo5.com by downloading one of the softphones and signing up. After you have completed the sign up process, be sure that you disable automatic startup of the softphone. You can't have your Asterisk system AND the softphone registering to the same Gizmo5 account!

Next, using FreePBX, Add a new Trunk named Gizmo5. For the Peer Details, insert the following using your actual Gizmo5 phone number and password:

type=peer
insecure=very
host=proxy01.sipphone.com
username=1747XXXXXXX
fromuser=1747XXXXXXX
fromdomain=proxy01.sipphone.com
secret=password
context=from-gizmo5-trunk
qualify=yes

Leave the Incoming Settings section blank and then enter the Registration String using your actual Gizmo5 phone number and password:

1747XXXXXXX:password@proxy01.sipphone.com

Save your settings and reload your dialplan when prompted.

Next, create a Default Inbound Route so that calls from Google Voice will be routed to extensions on your server. Then, create an Outbound Route called OutGizmo with NXXNXXXXXX and 1NXXNXXXXXX as the Dial Patterns and Gizmo5 as the main Trunk Sequence . Move this route to the top of your outbound routes to assure that U.S. calls are placed using the Gizmo5 trunk. Reload your dialplan when prompted.

Finally, log into your Asterisk server as root and insert the following lines at the end of extensions_custom.conf in the /etc/asterisk directory. Then reload the dialplan: asterisk -rx "dialplan reload"

[from-gizmo5-trunk]
exten => s,1,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => s,n,Goto(from-trunk,${DID_EXTEN},1)

Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. This forwarding phone number should be the DID that you want Google Voice to call when you have incoming calls on your Google Voice number. Again, to keep things simple, add your Gizmo5 phone number (747XXXXXXX) and select Gizmo as the Phone Type. You then will be prompted to place a test call and provide a 2-digit number to verify that the number is working. Answer the extension on your Asterisk system when it rings and enter the 2-digit code that's provided.

Gizmo5 Configuration. Log in to your Gizmo5 account using your 1747XXXXXXX account number or username and password. In the new Google Voice section of the form, insert your Google Voice email address and password. This is the email address you used to set up your Google Voice account. Choose "Use for U.S. calls only" and then click SAVE.

July 29 Update. Since this article was released, Gizmo5 has reduced the allowable calling time from unlimited to 20 minutes. Then today it was reduced to 3 minutes. That may be as long as you like to talk on the phone, but it's a major change from what was initially introduced 3 short days ago. Looks like we'll dust off our original article after all. Stay tuned...

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Deals of the Week. The nation's premier provider of free directory assistance service, 1-800-FREE-411, now is offering free 5-minute phone calls to most destinations around the world. Just listen to two quick commercials and enjoy your free call. Thanks, @MichiganTelephone. And now you can send free SMS messages worldwide from your iPhone. Thanks, @TruVoIP. Finally, AT&T has the refurbished 8GB iPhone 3G for $49 with a two-year contract.

Tweet

Originally published: July 26, 2009

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

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

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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.
 

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Some Recent Nerd Vittles Articles of Interest...

As many of you know, we were ecstatic when Google Voice arrived with free U.S. calling, voice mail transcription, and SIP connectivity to Asterisk®. That lasted less than a week until Google slammed the SIP door and spoiled our party. But... "Where there's a will, there's a way" goes the old adage, and so it is with Google Voice and free Asterisk calling. It returns today, back from the dead, thanks to another column of Gizmo tips and tricks that we wrote several months ago. Turns out our friends at Gizmo are considerably more persuasive with the Google moguls than we are because they still have SIP connectivity to Google Voice. That got us to thinking. If we have a free Gizmo account and a free Google Voice account, why can't we do a quick SIP marriage of convenience between the two accounts and restore free calling to Asterisk? Lo and behold, it turns out you can. And today we'll show you how. Admittedly, this isn't as exciting as other 3-ways you may have tried, but it's still fun! Once we get all the pieces in place, you'll be able to pick up any phone on your Asterisk system and place a call to anywhere in the United States for free. All you have to dial is GV + the area code and phone number. Couldn't be any easier!

Update: The original SIP interface to Google Voice described in this posting no longer works. A new approach that really works is now available on Nerd Vittles at this link.

Within the past few months, we've added several hundred million free phone numbers to our Asterisk PBX by creating a Skype Gateway as well as Gizmo Backdoor Dialing¹ and ENUM interfaces that didn't cost us a dime. When we add all the phones in the U.S. to that free calling list, we're getting very close to a billion free numbers. So welcome back... Googlified Messaging™.

Today's New Design. Much of today's column is a cut-and-paste job from previous tutorials because we really are marrying two previous tutorials to get this working. Here's the design. You need both a free Google Voice account and a Gizmo5 account. It's also free or almost free. Then we'll set up the Google Voice account to forward all incoming calls to voicemail. This allows you to call your Google Voice number and press a few keys to make an outbound call instead of listening to your voicemails. Next we'll set up our Gizmo5 account to forward all incoming calls to the SIP URI of our Google Voice number. We can get to Gizmo5 with a SIP call, but we can no longer place a direct SIP call to Google Voice. But Gizmo can! With a little dialplan voodoo, we'll tell Asterisk to place a SIP call to our Gizmo number. Gizmo then passes the call along with a SIP call to Google Voice. Asterisk counts to ten while the call is transferred to Google Voice. Then Asterisk acts like an auto-dialer by sending *, entering our Google Voice password, pressing 2, and finally dialing a 10-digit number plus # to place a free call to somewhere in the U.S. You'll never know any of this is happening behind the scenes until Aunt Betty answers her phone. And here's the best part of the story. The SIP call from your Asterisk server to your Gizmo number is free. The SIP connection from Gizmo transferring the call to your Google Voice number is free. And the phone call to any phone in the U.S. through Google Voice is free. FREE + FREE + FREE = FREE! And the sound quality is fantastic. The silver lining is that you can accomplish all of this and still use Google Voice as a message transcription service for your voicemails.

To get everything working, there are four steps: (1) configuring your Google Voice number to go directly to voicemail, (2) configuring your Gizmo5 number to forward all calls and send them via SIP to your Google Voice phone number, configuring FreePBX to route all calls with a GV prefix to your Gizmo5 SIP URI, and (4) configuring Asterisk to jump through the autodialer hoops to place an outbound call to any U.S. number through the Google Voice telephone interface. It sounds more complicated than it is. So hang on to your hat. Here we go!

Google Voice Design. To integrate free voicemail transcription and free U.S. calling into Asterisk, what we first must do is turn your Google Voice account into a glorified answering machine and message distribution system. When calls arrive on your Google Voice number, they will immediately trigger a greeting message that says something like this:

Thank you for calling Nerd Vittles. No one is available at the moment to take your call. After the tone, please identify yourself, leave a callback number, and a brief message. Your message will be transcribed and delivered to us. We will get back to you promptly. Please begin speaking after the tone.

Once a voicemail message is received, we want Google Voice to transcribe it and email us both the voicemail message and the transcribed text. The other feature we want is the ability to press *, enter our PIN, choose option 2 to place an outbound call, and dial a 10-digit number to any phone in the U.S. for free!

Google Voice Setup. Log into your Google Voice account and click Settings, General. In the Voicemail Greeting section of the form, record your greeting message as outlined above. In the Notifications section, identify the email and SMS addresses for delivery of your voicemail messages. In Voicemail Transcripts, check the option to transcribe voicemails. Now click on the Do Not Disturb check box to forward all inbound calls to voicemail.

Configuring Gizmo5. You can learn all about the host of features that Gizmo5 offers to the VoIP community by reading our previous article. Once you've set up your Gizmo account, log in and, in the Forwarding Gizmo5 Calls section of the form, click the Forward All Calls button. Next, click on the SIP option and enter your actual Google Voice phone number (instead of 6175171234) in SIP URI format: 6175171234@216.239.37.15:5061. Don't change anything else. Now click the Save button to save your settings.

Integrating Google Voice into Asterisk. This setup lets you place a call through Gizmo and Google Voice from any Asterisk phone by dialing the GV prefix plus a 10-digit number. So, to place a call to President Obama in Washington through Google Voice, you'd dial 48-202-456-1111. Good luck with that, but here's how...

17473456789 - Your Gizmo5 DID
8888 - Your Google Voice PIN

First, log into your Asterisk server as root and edit extensions_custom.conf. It's in the /etc/asterisk folder. Go to the very bottom of the file and insert the following code. Use your Gizmo phone number instead of 17473456789. Use your actual Google Voice PIN instead of 8888. Remember to expand the two-line dial string so it fits on a single line with no spaces! Save your changes and reload the dialplan: asterisk -rx "dialplan reload"

[custom-google-voice]
exten => _X.,1,Dial(SIP/17473456789@sipphone.com
,30,rD(wwwwwwwwwwwwww*www8888www2wwww${EXTEN}#))
exten => _X.,n,Hangup

Next, open FreePBX with a web browser and choose Setup, Trunks, Add Custom Trunk. Insert the following Custom Dial String on the form and Submit Changes and reload the dialplan:

local/$OUTNUM$@custom-google-voice

Finally, choose Setup, Outbound Routes, Add Route and fill in the following entries on the form:

Route Name: GoogleVoice
Dial Pattern: 48|NXXNXXXXXX
Trunk Seq: local/$OUTNUM$@custom-google-voice

Save your changes and reload the Asterisk dial plan one more time to complete the setup. Now you're all set to call the President whenever the urge strikes: 48-202-456-1111. And, remember, it's a free call... at least for now.

Creating Google Voice Favorites in Asterisk. If there are friends that you frequently call in distant places, you may find it more convenient to create Speed Dial numbers for them. Here's how to do it in Asterisk and still take advantage of free calling through Google Voice.

Log into your Asterisk server as root and again edit extensions_custom.conf in the /etc/asterisk folder. In the [from-internal-custom] context, add one or more entries for people you wish to call giving each of them their own extension on your PBX. Be sure to make the following substitutions and match your Gizmo and Google Voice credentials:

999 - Extension number to call
17473456789 - Your Gizmo5 DID
8888 - Your Google Voice PIN
1234567890 - Phone number of person to call

And here's the default entry which should be one continuous entry on one line:

exten =>999,1,Dial(SIP/17473456789@sipphone.com
,30,mD(wwwwwwwwwwww*ww8888ww2ww1234567890#))

When you finish making all the extension entries desired, save the file. Then reload your Asterisk dialplan: asterisk -rx "dialplan reload"

Google Dialer for Asterisk. Another approach for outbound calling with Google Voice would be to create a simple dialer in your Asterisk dialplan. The idea here is that anyone can pick up a phone and dial *GV (which is *48) to place a call. They then will be prompted to enter the 10-digit number to call. This code would be inserted in the same [from-internal-custom] context, and remember to insert your actual Google Voice PIN and Gizmo DID in the dial string. Keep the entire Dial command on a single line (which we can't do in this blog's template). Reload the Asterisk dialplan when you're finished.

exten => *48,1,Answer
exten => *48,n,Wait(1)
exten => *48,n,Set(TIMEOUT(digit)=15)
exten => *48,n,Set(TIMEOUT(response)=20)
exten => *48,n,Playback(pls-entr-num-uwish2-call)
exten => *48,n,Read(NUM2CALL,beep,10)
exten => *48,n,Playback(pls-wait-connect-call)
exten => *48,n,Dial(SIP/17473456789@sipphone.com
,30,mD(wwwwwwwwwwww*ww8888ww2ww${NUM2CALL}#))
exten => *48,n,Hangup

So... the ball is back in Google's court once again. Let's hope they make the right choice this time and leave SIP connectivity in place. Otherwise, S-K-Y-P-E is only a few small footsteps away. Enjoy!

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

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Some Recent Nerd Vittles Articles of Interest...

1. Backdoor Dialing unfortunately has bitten the dust at least temporarily. [↩]

Many of you may recall that last August we showed you an easy way to implement free calling to millions of cellphones using Gizmo5 and Asterisk®. Vaporware aside, it's been a quiet six months in the Skype for Asterisk department. But now the folks at Gizmo5 have outdone themselves once again. This time it's their new OpenSky service. Here's how it works. You can call as many Skype users as you like and talk for up to five minutes for free.

If there are people you frequently call and the time limit is a problem for you, then you can cough up $20 a year and make as many 2-hour Skype calls as you like to your ten best friends. If you've got more friends than that or if you plan to use this for something other than a home Asterisk system, then there are reasonably priced plans to accommodate you. $320 a year gets you 20 accounts to an unlimited number of Skype users with the same 2-hour per call limit. $800 buys you 50 accounts, and $1600 buys you 100 accounts per year.

Getting Started. The easiest way to integrate this into your existing Asterisk system is to sign up for a free Gizmo5 account and then follow our previous tutorial to set up your outbound trunk.

Once you have everything working, you're ready to add a few numbers on your Asterisk system for your Skype pals. Here's the easy way, and we'll cover some more sophisticated implementations in a subsequent article. Assuming you have a friend with a Skype username of joeschmo, here's what you need to do to call Joe by dialing 563 (J-O-E) from any extension on your Asterisk system.

Edit the /etc/asterisk/extensions_custom.conf file on your system and add the following line within the [from-internal-custom] context:

exten => 563,1,Dial(SIP/skype_joeschmo@proxy01.sipphone.com)


If you also use softphones which support SIP URI dialing, then you might want to add another entry like this in the same context:

exten => joeschmo,1,Dial(SIP/skype_joeschmo@proxy01.sipphone.com)


Now just reload your Asterisk dialplan, and you're ready to start calling your Skype buddies around the world from any Asterisk extension.

asterisk -rx "dialplan reload"


The FreePBX Alternative. As has been pointed out in a comment, you can accomplish much the same thing using newer versions of FreePBX without having to muck around in extensions_custom.conf. Just add an Extension, choose the Custom type, provide an Extension Number, a Display Name, and optional SIP Alias. Then insert the following in the dial field, save your entries, and reload the dialplan when prompted.

skype_joeschmo@proxy01.sipphone.com


$20 Buys You Skype Calling Aliases. One of the major drawbacks of Skype always has been the alphanumeric Skype names which make it next to impossible to place Skype calls using regular telephones. Well, Gizmo5 has solved that, too. With your $20 annual subscription which gets you 2-hour Skype calls to your 10 best friends for a year, you now can define new phone numbers to match against your 10 favorite Skype friends. For example, for a user named John Doe, you might choose 564-6363 (JOHN-DOE). Once you sign up for the $20 Skype subscription and configure this alias in your Gizmo account, you can reach John Doe on Skype by dialing 1-333-564-6363 through your Gizmo5 trunk from any Asterisk extension. In your Asterisk setup, just create an outbound route for Gizmo calls with the following dial strings, and you're all set.

1333NXXXXXX
333NXXXXXX


Special thanks to JPE on the PBX in a Flash Forum for the original tip and to Adrian at Gizmo5 Operations for the alias demo. Enjoy!

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Want a Bootable PBX in a Flash Drive? Our Atomic Flash bootable USB flash installer for PBX in a Flash has been quite the hit this past week. Thank you to all of our generous contributors! Atomic Flash provides all of the goodies in the VPN in a Flash system featured last month on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and this USB flash installer in less than 15 minutes!

If you'd like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment with your best PBX in a Flash story.¹

Be sure to include your real email address which will not be posted. The winner will be chosen by drawing an email address out of a hat (the old fashioned way!) from all of the comments posted over the next several weeks.

And it's still not too late to make a contribution of $50 or more to the PBX in a Flash project and get a free Atomic Flash installer delivered to your door as our special thank you gift. See this Nerd Vittles article for details.

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New Fonica Special. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. For outbound calling, we recommend you establish accounts with several providers. We've included two of the very best! These include Joe Roper's new service for PBX in a Flash as well as our old favorite, Vitelity. To get started with the Fonica service, just visit the web site and register. You can choose penny a minute service in the U.S. Or premium service is available for a bit more. Try both. You've got nothing to lose! In addition, Fonica offers some of the best international calling rates in the world. And Joe Roper has almost a decade of experience configuring and managing these services. So we have little doubt that you'll love the service AND the support. To sign up in the USA and be charged in U.S. Dollars, sign up here. To sign up for the European Service and be charged in Euros, sign up here. See the Fonica image which tells you everything you need to know about this terrific new offering. In addition to being first rate service, Fonica is one of the least expensive and most reliable providers on the planet.
 
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.
 

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Some Recent Nerd Vittles Articles of Interest...

1. This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. [↩]

We're wrapping up our Gizmo5 series with what we believe is the real silver lining in the Gizmo Project. Here's our YouTube demo to prove it. We kicked things off by showing you how to set up a Gizmo5 account to make free calls with Asterisk® using Backdoor Dialing and ENUM. Then last week we added an Asterisk dialplan script to actually test whether an outbound call would be free through Gizmo5 before choosing a final route to terminate outgoing calls from your Asterisk server. Today we're going to use your Gizmo5 account to transform a standard Nokia N95 cellphone into a SIM-free, carrier-free WiFi SIP telephone which can perform a whole bag of tricks at absolutely no cost... once you own the unlocked phone. It's the perfect back-to-college gift if your wallet matters. Without too much hype, suffice it to say the N95 earned PC Magazine's Editor's Choice. In a word, the call quality is phenomenal. If you like Snickers candy bars, then you'll love the size of this phone. And WiFi works all day with Nokia's Symbian OS even though you're lucky to make it an hour with Windows Mobile devices. Maybe it's not the WiFi that's the problem after all, Bill. Ooops. He retired. Sorry. Anyway, any unlocked Nokia N95 will do. Just not the N96! And you'd better get one while they last. Nokia apparently has had a change of heart on SIP telephony support, and it's quickly disappearing from their newer models. Dumb move!

SPECIAL NOTE: We have one, gently used N95 for sale. It actually was used to prepare this article. Make us an offer, or we'll make you a deal you can't refuse. If you're interested, contact us.

When we're finished, you'll have a better appreciation for why AT&T and the other cellphone carriers hate Nokia phones and why Comcast would prefer to limit your bandwidth and charge you $40 (extra) per month for their VoIP service and boatloads more for their pay-per-view movies. This isn't about greedy bandwidth abusers. It's about a greedy service provider. Comcast could easily rein in bandwidth abusers with a letter threatening to terminate service. What they can't control with nastygrams are the Blockbuster's and Walmart's of the world that want to deliver pay-per-view movies to your doorstep via the Internet. So this looks more like restraint of trade to us than protection of scarce resources from the Napster generation as Comcast would have you believe. You don't make the whole class stay after school because one kid chewed gum... if your motives are as pure as the Comcast TV barons would have you believe. Now where were we? </rant>

For openers today, you can place SIP calls at no cost to any SIP phone or Asterisk server in the world. See our previous tutorial to learn how to set this up on your Asterisk server. Second, you can place "regular" phone calls to any phone in the world using a Gizmo5 account at Gizmo5's discounted calling rates (2¢ a minute or less for calls to U.S., U.K., most of Europe, China, and Australia for example) rather than cellphone subscription plus stratospheric long distance charges. Third, you can place free calls to almost every non-AT&T cellphone in the U.S. by dialing 0101 and the number. Fourth, you can place free calls back to and through your Asterisk server to just about anywhere on the globe (except resort areas surrounded by water) for almost nothing. And, finally, you can receive free calls on your N95 cellphone whenever anyone dials your free IPkall-assigned DID in the Seattle area or your SIP number through Gizmo5. And, did we mention that all of this magic occurs with no connection to AT&T or any other cellphone carrier. In fact, we don't even have a SIM chip in our N95. Well... not all the time anyway. All you really need is a WiFi connection to make all of this work. And even the Asterisk server is optional. So let's get started.

Enabling WiFi on Your Nokia N95. Before we can use the N95 as a free SIP telephone, we've first got to get a WiFi connection enabled on the phone. Pressing Menu, Tools, WLAN Wizard will get you started. You can test your connection by opening the web browser for a trial run after you have your WiFi connection set up. Once it's working, be sure to disable the WiFi Access Point scanning feature by choosing Menu, Tools, Settings, Connection, Wireless LAN and set Show WLAN availability to Never.

Installing the Gizmo5 Application. Now the tricky part, and it's really not that difficult. It just happens that there's lots of conflicting information posted around the web, and this makes the drill more confusing than it needs to be. First, if you already have automatic registration of your Gizmo5 account on another device or an Asterisk server, disable the automatic registration. You can't have the same account registered in two places simultaneously. Just open a second account if you need it. There are two components that need to be installed on your Nokia N95, and they're in different places. First, install Nokia Internet Services Support Package to the device memory (not to the memory card). Here's Nokia's download link. Next, install the Gizmo5-Nokia PlugIn from gizmovoip.com. Here's the download link for that one. Finally, we had one little gotcha with getting everything to work once it was installed. On your phone go to Menu, Tools, Settings, Connection, SIP Settings, Options, Edit SIP Profile and set the Service Profile to Nokia 3GPP. Next, go to Menu, Tools, Internet Tel and activate Gizmo after choosing your default WiFi network. You'll be prompted for your Gizmo account name and password. Once it's registered, you should be able to dial 0101 plus an area code and phone number to test out the free calling feature. Or you can dial an area code and number, and route your outbound call as a pay-by-the-minute Gizmo5 Internet Call under the Options button. To call a sip phone directly, simply create a new Contact and insert an Internet telephone entry in the SIP URI format: sipname@FQDN.com. Once you have saved the entry, simply choose it from your Contacts to place the free SIP call. In Nokia-speak, it's referred to as an Internet Call.

If the above procedure doesn't work for you, repeat the drill and set the Service Profile to IETF instead of Nokia 3GPP. Not sure why but one setting works some of the time, and the other one works the rest of the time. If you can't connect, this is usually the problem... assuming you've gotten your Gizmo5 username and password entered correctly.

You also can use your Asterisk server to forward outbound SIP calls from your N95 to other phones. For example, if there are 10 close friends that you call frequently, assign each of them a SIP URI on your Asterisk server. We covered the setup process in this article. In a nutshell, create an Incoming Route in FreePBX named tom and point it to the phone number you wish to call. For destination phones outside your PBX, first create a Miscellaneous Destination called Tom-home that includes the home phone number. Then use this destination in your Incoming Route for tom. Save your entry and reload your FreePBX dialplan. Finally get your own fully-qualified domain name from a service such as dyndns.org. Assuming your FQDN was pbx.dyndns.org, then your Internet telephone entry for Tom in your N95 contacts would be this SIP URI: tom@pbx.dyndns.org. Otherwise, you'll need a SIP URI with the IP address of your Asterisk server, e.g. tom@36.24.36.1.

Adding a Free DID for Inbound Calling to Your Nokia N95. One of the world's best kept secrets continues to be the availability of free DIDs from ipkall.com in Seattle. This saves you $35 a year over the current Gizmo5 DID rate, and IPkall will give you a free phone number in one of several available area codes to use with the SIP device of your choice. Your Nokia N95 qualifies! Just be sure to place at least one call a month to the number, or it's automatically recycled to someone else. To register for a free IPkall account, go to this link and sign up. Your SIP Phone Number is your 11-digit Gizmo5 phone number starting with a 1. Your SIP Proxy for Gizmo5 is proxy01.sipphone.com. Now plug in a valid email address and create a password for your account. Your new phone number will be delivered to this email address. Once it arrives, you should be able to dial the number from any phone, and your Nokia N95 should start ringing. Answer the call just as you would any other cellphone call. The only difference is that you can talk as long as you like... for free. For other free DIDs and some great tips including ATA setup, go here.

Using Asterisk to Add the Missing Pieces. There are a number of ways you can use Asterisk to enhance your SIM-free Nokia experience. By enabling DISA, you can place a SIP call to your Asterisk server, obtain dial tone, and call anywhere using your existing Asterisk trunks. Here's the way we set this up. Edit /etc/asterisk/extensions_custom.conf and add a [custom-disa] context at the end of the file that looks like the following code. Be sure to set a VERY secure password in line s,7 by replacing 1234. It's your phone bill! Then set your IPkall DID number as the CallerID in s,13. By changing 701 in s,12 you can call any extension on your Asterisk server just by dialing 0 when you're using DISA. For our foreign friends, be sure to adjust the dial string length (10) in s,9 to meet your local needs.

[custom-disa]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Set(TIMEOUT(digit)=7)
exten => s,4,Set(TIMEOUT(response)=10)
exten => s,5,Background(enter-password)
exten => s,6,Read(MYCODE,beep,7)
exten => s,7,GotoIf($["${MYCODE}" = "1234"]?8:15)
exten => s,8,Set(TIMEOUT(absolute)=9000)
exten => s,9,Read(NUM2CALL,pls-entr-num-uwish2-call,10)
exten => s,10,Playback(pls-wait-connect-call)
exten => s,11,GotoIf($["${NUM2CALL}" = "0"]?12:13)
exten => s,12,Dial(Local/701@from-internal)
exten => s,13,Set(CALLERID(number)=4251234567)
exten => s,14,Goto(outbound-allroutes,${NUM2CALL},1)
exten => s,15,Hangup


Next, add an Incoming Route using FreePBX. For the DID Number, use a SIP name that is not easily guessed, e.g. DISA2375. This gives you an extra layer of password protection since anyone can try to guess your SIP URI's once they know the IP address of your Asterisk server. Leave all of the other entries at their defaults and, for the Destination, choose Custom Route: custom-disa,s,1. Save your settings and reload your dialplan. Ignore the warning that you're doing something odd. We know what we're doing.

Finally, on your Nokia N95, add a new Contact called DISA with an Internet telephone number to match the name of your incoming route above with the fully-qualified domain name of your Asterisk server, e.g. DISA2375@pbx.dyndns.org. Now you're ready to dial away by simply selecting this contact on your N95. Enter your DISA password when prompted and then enter a 10-digit phone number to call.

The WiFi HotSpot Two-Step (and a few more steps). Now that everything is working swimmingly, we're ready to take your Nokia N95 on the road. Here's the failsafe step-by-step to get connected in a WiFi HotSpot of your choice.

• Turn off the phone
• Arrive at HotSpot
• Turn on the phone
• Menu, Tools, WLAN Wiz., Pick Your HotSpot, and Start Web Browsing, Create WLAN While OffLine=Yes
• Using Browser, log into the HotSpot with your account name and password
• Leave the browser open so it'll be easy to log out when you're finished
• Menu, Tools, Internet Tel., Pick Your HotSpot AP
• Once Connected, Dial Away As Usual
• When finished, Hold Down Menu Button and Choose Browser App, Log Out of the HotSpot
• Turn off the phone

Cellphone Options. But what if you really do want to use the Nokia N95 in all its glory with the 5 megapixel camera and the multimedia goodies and even a cellphone provider? Well, it works great for just about anything you need. In fact, you can even take the SIM chip from your iPhone (even a First Generation iPhone) and plug it in. Phone calls work, voicemail works (even though you get two text messages when a new voicemail arrives... which is a lot better than Cingular in the old days when you typically got zero), and email and web browsing work great, too. Just select MediaNet as the access point when you open your Internet connection, and you'll be off to the races. Of course, all the cheapo, pay-as-you-go SIM cards work as well. Both Oxygen and Airvoice packages including free minutes and a SIM card are available at this eBay store for under $10. And there are lots of other options as well. Enjoy!

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VPN in a Flash Update! We've had over 100 reservations for our new VPN in a Flash system. We're very close to having a manufacturer in place so hopefully we'll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It's not too late to get in the queue and place a reservation for a system. Just send us a note, and we'll keep you posted as the release date approaches. It'll hold your place in line with absolutely no obligation to purchase.

Coming Attractions. We're very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada. And a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone is just around the corner. Stay tuned!

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Hosting Provider Mega Deal. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn't get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.

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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. Until October 15, you 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.

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Some Recent Nerd Vittles Articles of Interest...

Last week we introduced a couple of new free calling options for Asterisk®: ENUM and Gizmo5's Backdoor Dialing. But one of the limitations of the Gizmo5 service in particular was the need for a 0101 prefix in order to trigger a free call as opposed to a pay per minute call to the same number. This highlighted a pretty serious limitation in the way FreePBX processes most outbound calls. As we indicated, the process goes something like this. After a caller dials a number, FreePBX searches through its Outbound Routes (from top to bottom) looking for a match on the dial string. Once it finds one, FreePBX then initiates calls beginning with the first trunk in the trunk priorities list for that outbound route. If the call is completed, no further call processing takes place. And a completed call includes a call that is either answered or rings busy. If a call is not completed, FreePBX continues to drop down the available trunks list and repeats the process until a call is either completed or the trunk list is exhausted. The one exception to this scenario was support for ENUM. In that situation, a lookup occurs after a call is dialed to see if it can be placed as a free SIP call. We'd like to do the same thing with Gizmo5's Backdoor Dialing. What we want to do is query the Gizmo5 database to determine whether a number to be called is a free call. If it is, then we want to modify the route for processing the outbound call to take advantage of Gizmo5's free calling option. And we'll also need to change the phone number by adding a 0101 prefix.

Since our last column, another serious limitation in FreePBX post-call processing was mentioned on the PBX in a Flash Forums. With a number of commercial PBXs, it's possible to specify post-dial processing for emergency calls. For example, in an office environment, if an employee dialed 911, it would be helpful to alert a receptionist in some way so that immediate first aid could be attempted and also to give the receptionist a heads up so that he or she could direct emergency responders to the appropriate location in a building. As written, FreePBX doesn't provide an easy way to handle this.

So our objective today is to provide a couple applications which address these limitations. And the apps also will document a methodology for overcoming other post-dial processing issues which may arise using the existing FreePBX framework.

The trick to adding today's hooks into the Asterisk dialplan is to understand that Asterisk loads identically named dialplan contexts only once. Taking advantage of this, FreePBX provides a mechanism for users to insert custom code to replace default FreePBX contexts. All of these configuration files are stored in /etc/asterisk. For today, the context we want to modify is [macro-dialout-trunk]. This is the FreePBX macro that does the heavy lifting once a call has been placed and a trunk route has been selected to handle the call. With FreePBX 2.3, the macro is in extensions.conf. In FreePBX 2.4 and 2.5, the context is in extensions_additional.conf. In both cases, what we want to do is copy the entire contents of the existing context into the bottom of extensions_override_freepbx.conf. If you're using an editor to cut-and-paste the code, be sure you get the code that is located outside the left and right margins of your editor. And the context ends on the line before the next context begins. In the case of FreePBX 2.3, the next context is [macro-agent-add]. In the case of FreePBX 2.4, the next context is [macro-dialout-dundi]. And, in 2.4, there is now a comment which indicates where each context ends: ; end of [macro-dialout-trunk].

What we want to do is insert a line or two of custom code in this context which you've copied into extensions_override_freepbx.conf. The purpose is to run our custom code after the number to dial and trunk ID have been passed to this macro. Then, in the case of the Gizmo5 application, we'll run out to the Internet to determine if this call should be handled as a free call. If so, we'll change the trunk ID number to match your Gizmo5 trunk, and we'll change the number to dial by prefixing the existing number with 0101. The only gotcha with the Gizmo5 Backdoor Dialing is that every number must be tested at least once by someone (not necessarily you) in order to populate the Gizmo5 free calling database. You can check as many numbers as you like at this link. In the case of our 911 emergency application, we'll check to see if the number being dialed is 911. If so, we'll send an email or text message to an address that you define with an alert that extension 1234 just placed a call for emergency assistance to 911.

If you're using FreePBX 2.3, the custom code below should be inserted after the third "exten" line in the context, i.e. after the following line of code:

exten => s,n,Set(ROUTE_PASSWD=${ARG3})


If you're using FreePBX 2.4, the custom code below should be inserted after the first "exten" line, i.e. after the following code:

exten => s,1,Set(DIAL_TRUNK=${ARG1})


And the code to be inserted looks like this for Asterisk 1.4:

exten => s,n,AGI(nv-outbound.php|${ARG2}|${ARG1})
exten => s,n,AGI(nv-gizmo.php|${ARG2}|${ARG1})


For Asterisk 1.6, it should look like this:

exten => s,n,AGI(nv-outbound.php,${ARG2},${ARG1})
exten => s,n,AGI(nv-gizmo.php,${ARG2},${ARG1})


Now we need to add a couple of PHP scripts to your system and set a few configuration options, and you'll be ready to go. While logged into your server as root, issue the following commands:

cd /var/lib/asterisk/agi-bin
wget http://pbxinaflash.net/source/gizmo/nv-gizmo.zip
unzip nv-gizmo.zip
rm nv-gizmo.zip
wget http://pbxinaflash.net/source/gizmo/nv-outbound.zip
unzip nv-outbound.zip
rm nv-outbound.zip
chown asterisk:asterisk *.php
chmod +x *.php
asterisk -rx "dialplan reload"
grep OUT_ /etc/asterisk/extensions_add* | awk '/ = / { print $0 }'


The last line of code above is used to decipher the trunk numbers associated with each of your trunks. What we need to know is the trunk number for the Gizmo5 trunk that you set up in last week's tutorial. Write it down and then edit nv-gizmo.php: nano -w nv-gizmo.php. Look down the screen about 5 or 6 lines for the line that reads $GIZMO_TRUNK = "21" ; and replace 21 with the number you wrote down for your actual Gizmo5 trunk. In the next two lines, insert your actual Gizmo5 username and password between the quotation marks. Don't change anything else. Save your changes: Ctrl-X, Y, and then press the Enter key.

With the other application, nv-outbound.php, we need to be sure it's working with your phone system before you actually need it. And we don't place test calls to 911. So here's the drill. Edit the file: nano -w nv-outbound.php and insert your email address or text message address in the $email variable between the quotes. Then move to the next line and insert a telephone number with the area code that you can dial from a phone on your system to test that the notification is working. For example, put in your cell phone number. Once you save your changes, pick up a phone on your system and call your cellphone. You should receive an email notification within a few seconds. Once it's working, edit the application again and change the $number2monitor to "911" and you're all set. Enjoy!

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VPN in a Flash Update! We've had over 100 reservations for our new VPN in a Flash system. We're very close to having a manufacturer in place so hopefully we'll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It's not too late to get in the queue and place a reservation for a system. Just send us a note, and we'll keep you posted as the release date approaches. It'll hold your place in line with absolutely no obligation to purchase.

Coming Attractions. We're very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada as well as all-you-can-eat plans for just over $10 a month with an annual contract. We're also only a week or two away from a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone. Stay tuned!

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Hosting Provider Deal of the Century. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn't get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.

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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. Until October 15, you 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.

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Some Recent Nerd Vittles Articles of Interest...

It’s been a while since there’s been much to cheer about in the free calls department with Asterisk®. But today, to kick off the new school year, we have lots of good news and some simple tricks to add zillions of free phone numbers to your Asterisk repertoire. In fact, you’ll be able to call almost any non-AT&T cellphone or landline in the United States at no cost. Remember that when you buy your next cellphone! Special thanks to Cliff on the PBX in a Flash Forums for heads up.

Some early readers of Nerd Vittles may remember sipphone.com which morphed into Gizmo5.com. In January of this year, Gizmo5 struck peering deals with a number of telephone providers that already routed their calls over the Internet. And it’s a pretty impressive list that includes more than 10% of the phones and cellphones in the United States according to Gizmo5′s bean counters. There’s Access One, Airadigm, Allegiance, Alltel, Cablevision Lightpath, Cat Communications, Cbeyond, Cellcom, Cellular Properties, Centennial Wireless, Choice One, Cincinnati Bell Wireless, Cinergy Communications, Cingular, CityNet, Cleveland Unlimited, Comcast Digital Voice, Commpartners, Conversent Communications, Cox Communications, CP Telecom, CTC Communications, Dobson Cell, Eureka, Globalcom, Heartland Communications, Illinois Valley, ITC Deltacom, LDMI, McLeod, Metro PCS, Mpower, Nationsline, Nextel, Nextera Communications, Paetec, RCN, Sprint PCS, Talk America, Telnet Worldwide, T-Mobile, US Cellular, Verizon Wireless, and XO. Whew! And the program is constantly being expanded. Toll-free numbers and Gizmo5-to-Gizmo5 calls also are free using Gizmo5. You can check whether your frequently called numbers are free calls by simply entering the phone numbers at this link.

Thus was born what Gizmo5 calls Backdoor Dialing. Just dial 0101 and the 10-digit number of your choice. If it’s free, the call goes through. If not, you get a message that the number is not yet supported and click. The beauty of the program is that your total investment to use the free service with Asterisk is a one-time fee of $10 for a bucket of CallOut minutes to activate your account. Sometimes this takes a day for the credit to appear, particularly if you use PayPal to cover the cost. The good news is you can spend most of the $10 making calls to any phone in the world, many for under 2¢ per minute, using just about any computer on the planet. Just leave a few cents in the pot to keep your free Backdoor Dialing service enabled. From our testing, we’d rate the Gizmo5 call quality as excellent on both the free and the pay-per-minute calls! Complete rate tables are available here.

Gizmo5 provides free softphones for Windows, Macs, and Linux as well as numerous cell phones and mobile devices including Treo, Nokia, and many more (not the iPhone… yet!). All of the softphones make it extremely easy to place SIP calls, e.g. joeschmo@mypbx.dyndns.org. And you can place these calls all day long at no cost. See our tutorial for step-by-step instructions on setting up your own SIP addresses on your Asterisk server. The softphones also include Conferencing, SMS, and Instant Messaging with AIM, Yahoo, MSN, Google, and MySpace.

As with many of these services, they weren’t designed for Asterisk, but nothing in their fine print precludes Asterisk use so today we’ll show you how. Will the program last forever? Who knows, but it’s free for now. And the cost of admission is too good to resist. You’re obviously not going to dial every number you frequently call twice just to see if the call is free. That’s why you’ll want to use a robodialer such as AsteriDex for your outbound calling. Then it’s easy to adjust the phone numbers of your friends with Sprint, T-Mobile, or Verizon cellphones so that you never have to pay for those calls again. Just add a prefix of 0101 to the numbers, and you’re done. And they can call you on your Gizmo5 CallIn number through Asterisk if you’ve enabled the CallIn Service and chosen a number. It’s under $3 a month with an annual subscription. Or the calls can be returned using the CallerID number displayed by Gizmo5 when you call your friends. Toll charges may apply in this case due to the Gizmo5 area code.

So let’s get started. Step 1 is to download and install a free softphone of your choice and follow the prompts to sign up for your account. There’s really no reason not to install a Gizmo5 softphone on every computer you own. If you don’t use it, there’s no cost. If you ever need it, it’ll be there for you. Step 2 is to make a $10 purchase of CallOut minutes. While you’re waiting on the credit to appear (and it usually takes less than a day), let’s set up Asterisk. You’ll need your new account name, password, and phone number from Gizmo5 to get started.

Setting Up a FreePBX Trunk for Gizmo5. If you’re using a product such as PBX in a Flash that includes FreePBX, then open FreePBX in your browser and choose Setup->Trunks->Add SIP Trunk. Leave the General Settings blank. For the Dialing Rules, if you just want free calling through your Gizmo5 trunk, plug in values below. For regular calls as well, add 1NXXNXXXXXX or an entry that is suitable for each country you wish to call.

1800NXXXXXX
1822NXXXXXX
1833NXXXXXX
1844NXXXXXX
1855NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
800NXXXXXX
822NXXXXXX
833NXXXXXX
844NXXXXXX
855NXXXXXX
866NXXXXXX
877NXXXXXX
888NXXXXXX
0101+NXXNXXXXXX
0101NXXNXXXXXX

Name the Trunk: Gizmo5. Make the following entries in Outgoing Settings Peer Details:

disallow=all
allow=ulaw
auth=md5
authuser=youracctnameNOTyourphonenumber
canreinvite=no
context=from-trunk
dtmfmode=auto
fromdomain=proxy01.sipphone.com
fromuser=youracctnameNOTyourphonenumber
host=proxy01.sipphone.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
type=peer
username=youracctnameNOTyourphonenumber


Clear out the Incoming Settings and use the following syntax for the Registration String. Then Save your setup and Reload Your Dialplan. NOTE: Don’t use any registration string unless you want incoming call support. By not registering, you can use your softphones whenever you need it to also make outbound calls. If you register with Gizmo5 using a registration string, then it knocks out use of a softphone since you can’t have two simultaneous registrations to the same account. But registering allows those you call with this service to call you back conveniently… although not necessarily for free from the caller’s phone.

youracctname:yourpassword@proxy01.sipphone.com/yourphonenumber


Setting Up a FreePBX Outbound Route for Gizmo5. While still in FreePBX, choose Setup->Outbound Routes->Add Route. Name the route: OutGizmo5. Then enter the following Dial Pattern: 0101NXXNXXXXXX. Choose SIP/Gizmo5 as your Trunk Sequence. Then click Submit Changes and Reload Your Dialplan.

Setting Up a FreePBX Inbound Route for Gizmo5. While still in FreePBX, choose Setup->Inbound Routes->Add Incoming Route. Name the route: Gizmo5 and plug in your 10-digit DID number in the appropriate field. Then Set a Destination for the incoming calls. That’s it. Save your entries by clicking the Submit button and then Reload Your Dialplan.

Making a Free Call with Gizmo5. Once your DialOut credit appears on your softphone or in your Gizmo5 web account, you’re ready to start making calls. From any phone connected to your Asterisk server, just dial 0101 plus the 10-digit phone number. On the Asterisk CLI, you should see the call routed out through your SIP/Gizmo5 trunk. If you get a congestion tone and you’re sure your DialOut credit has been posted to your account, then check your username and password entries in your Trunk setup. Be sure to use your account name and NOT your Gizmo5 phone number for your username, authuser, and fromuser entries. But, if that doesn’t work, try using your Gizmo5 phone number instead of your assigned user name. Some have reported quirks in which actually works. For us, the assigned user name did the trick. Also make certain that the disallow all entry is above the allow=ulaw in versions of FreePBX after 2.3, or no calls will ever be successful.

Turning Non-Free Numbers into Freebies. There’s always some enterprising individual that figures out a quick way to beat the system even when many calls already are free. Suppose the number you wish to call isn’t yet available through Backdoor Dialing. The only trick is to have a pool of numbers from a provider with a peering arrangement with Gizmo5… and, of course, an Asterisk or FreeSwitch server to forward the calls and handle the number translation. You can read about RingBranch’s implementation, and then you can sign up for the service here.

There’s another way to turn non-free calls into freebies. This is Gizmo5′s "All Calls Free" Plan which is available in 60 countries. Landlines and mobile phones are supported in 17 countries while landlines only are supported in 43 more. U.S., Canadian, and Chinese landlines and cellphones are included in the program in addition to those of the Pope and the other residents of Vatican City. God works in mysterious ways! Here’s the complete list of countries that are supported.

To qualify a landline or mobile number for free calling (by dialing with the usual country code prefixes), you both have to be "active" Gizmo5 subscribers, your landline and mobile numbers must be listed on your account, and you must enter each other in your respective Buddy Lists. Then free calls using your Asterisk Gizmo trunk can be made to the "regular" phone numbers of all your pals whether the called person is online with Gizmo or not. Be aware that you can’t call your own numbers for free, and there is lots of additional "fine print" in this program. Nothing precludes your spouse having his or her own Gizmo5 account, however. You’ll need to wade through the rules carefully to take advantage of the free calling. It is possible, but it’s not easy. If you have relatives in Europe, Australia, or the Far East, you might want to have a look here. Just do a search for "All Calls Free." Your Gizmo5 softphone also will report your current All Calls Free Status.

Add Free Calls to 40 Million Asterisk Servers with e164.org. While we’re on a roll of free calling, here’s a simple way to add free calling to 40 million Asterisk servers around the world. Just add your name and phone numbers to the e164.org registry at no cost and configure FreePBX with ENUM support. Then outbound calls to numbers in the e164 registry will always be free as well. The whole setup takes less than 10 minutes. Here’s how.

The first step in setting up ENUM is to create a SIP address for your Asterisk server. The format looks like this: myname@somedomain.com. You’ll need either a fully-qualified domain name (FQDN) if your server has a static IP address or an FQDN issued through a dynamic DNS service such as dyndns.org if you have a dynamic IP address, e.g. pbx.dyndns.org. In the latter case, your router keeps dyndns.org apprised of changes in your external IP address so that pbx.dyndns.org always resolves to the correct IP address of your Asterisk server. Incidentally, with any hosted domain using a registrar such as omnis.com, it’s easy to add a subdomain DNS entry and point it to your Asterisk server, e.g. sip.joeschmo.com. That won’t cost you a dime other than the annual $6.95 domain registration fee which you’re already paying anyway.

Step two is to add your new FQDN address with a name of your choice to your Asterisk server. Then Asterisk will know how to process incoming SIP calls to that address. Read the Rolling Your Own section of our article on SIP Proxies for the procedure using FreePBX. It only takes a minute or two to set up. Let’s assume for purposes of this tutorial that you’re going to use the following destination address on e164.org for your server: e164@pbx.dyndns.org. An advantage to this type naming scheme is you can always keep straight the source of your incoming SIP calls. Thus your /etc/asterisk/extensions_override_freepbx.conf file should include a line in the [from-sip-external] context that looks like this: exten => e164,1,Goto(from-trunk,e164,1)

This tells Asterisk to route incoming SIP calls to e164@pbx.dyndns.org to the FreePBX Incoming Route for e164. And to complete the routing of the inbound calls to this address, add an Inbound Route in FreePBX called e164 that includes a destination of your choice for these SIP calls, e.g. an extension, a ring group, or an IVR already configured on your system. Just a footnote that e164.org requires you to enter a confirmation PIN when you set up the SIP routing to your server. So, at least initially, make the destination for your e164 SIP calls an extension that you can answer to obtain your PIN. You can safely ignore the FreePBX warning that you’re entering an odd type of inbound route by clicking OK. But you knew that.

Now let’s get you signed up with an account on e164.org. Go to the web site and click the Sign Up tab. Go through the sign up drill and then log into your new account. Then click the Phone Numbers tab and Add your phone numbers to e164. For each number, enter the area code and number. Then click the Next button. You’ll be warned about not having the number you’ve specified redirected to an IVR. If you already have this DID redirected to an IVR, change the routing temporarily to an extension that you can answer to obtain your PIN before you press Next to proceed. You’ll then be prompted for the SIP address to contact your server. Leave the default SIP protocol and plug in the address you created, e.g. e164@pbx.dyndns.org (using your own FQDN, of course). As soon as you click the Next button, your phone should start to ring, but there may not be a message when you answer. Hang up and wait for the second call within 15 minutes. It will include your PIN. Now click on the Phone Numbers tab and update your phone entry by choosing Enter PIN and typing your assigned PIN. Your phone number now has been activated with the e164 service. To complete the setup, you’ll want to click on the Do Not Call option and make your selections. You also can decide whether to list yourself in the ENUM White Pages directory.

Remember that the real purpose of this drill was to avoid charges when you place outbound calls to numbers in the ENUM directory. We merely added your numbers to e164.org so that others could benefit as well. So the final step before you can start saving money is to configure FreePBX to handle ENUM lookups for outbound calls from your server. One more observation may be helpful. You’ll recall that one of the limitations of FreePBX has always been that once an outbound route was chosen for a call, if the call was completed using the first destination trunk in that route, then the call processing ended there. ENUM adds a new wrinkle because we basically want to connect to ENUM to check for a free route and, if no matching entry is found, then we want the next trunk to process the call. As luck would have it, FreePBX has been tweaked to allow this scenario. All you have to do is create an ENUM trunk and then place it first in your sequence of trunks for each of your outbound routes. If an ENUM entry is found for the number you’re calling, the call will be routed as a free call with a direct SIP connection. Otherwise, the call processing will continue and the call will be routed using the next trunk specified in your outbound route.

There are two steps in FreePBX to implement ENUM. First, we need to create a special ENUM trunk. And second, we need to adjust our outbound routes to use the ENUM trunk first, and then the series of trunks you already have specified in each outbound route. NOTE: You obviously wouldn’t do this for an emergency 911 outbound route.

In FreePBX, click Setup, Trunk, Add ENUM Trunk. Enter your desired CallerID for these calls. Set a maximum number of channels, if desired, and then leave the other entries blank in most cases. Save your settings and reload your dialplan. Now click Setup, Outbound Routes and adjust the sequence of trunks for each of your existing routes. Be sure to put ENUM in the top position of each desired route. We also recommend adding a new Free Calls route so that users on your system can dial 0 and then a number to place a call through ENUM and then Gizmo5. If neither has a route for calling the party for free, the call will fail. The dial patterns might look like this for U.S. calls:

0|1NXXNXXXXXX
0|NXXNXXXXXX

The trunk list would look like this:

0 ENUM
1 SIP/gizmo5


Continue reading Part II.

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Today’s Must Read: 101 Things You Can Do With Asterisk

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VPN in a Flash Update! We’ve had over 100 reservations for our new VPN in a Flash system since last week. We’re very close to having a manufacturer in place so hopefully we’ll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It’s not too late to get in the queue and place a reservation for a system. Just send us a note, and we’ll keep you posted as the release date approaches. It’ll hold your place in line with absolutely no obligation to purchase.

Coming Attractions. We’re very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada as well as all-you-can-eat plans for just over $10 a month with an annual contract. We’re also only a week or two away from a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone. Stay tuned!

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Hosting Provider Deal of the Century. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn’t get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.

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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. Until October 15, you 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.

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