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JUST RELEASED: Visit the Incredible PBX WikiTweet2Dial: SMS Messaging with Google Voice and Twitter
We continue our quest for convergence today by adding the missing piece to our recent Tweet2Dial application. In addition to free calls to everyone in the U.S. and Canada as well as complete management of your Asterisk® server from Twitter, today's enhancement lets you send SMS messages to any SMS device or cellphone in the U.S. and Canada using simple Twitter messages. And, best of all, Tweet2Dial is free and runs on almost any Asterisk or Linux server as well as every Mac on the planet.
Twitter already provides some basic SMS integration that allows you to use SMS messages to send tweets. You also can opt to receive some Twitter messages via SMS whenever your friends post a new Tweet. But Twitter's SMS functionality is Twitter-centric meaning that both you and your friend must be Twitter users to take advantage of the SMS enhancements. Tweet2Dial adds the missing piece so that you can send SMS messages to anyone with an SMS-capable device in the U.S. and Canada whether or not they have a Twitter account. After all, that's what convergence is all about!
If you've already installed Tweet2Dial, we'll walk you through upgrading your existing setup in this article. If you haven't previously installed Tweet2Dial, then all you need to do is read the updated, original article which now includes coverage of the SMS functionality. Keep in mind that current Twitter API call limitations still limit you to one call or SMS message or Asterisk CLI command per minute. We'll remove this limitation once Twitter expands the hourly API call restriction.
Upgrading Tweet2Dial. For those that already have installed Tweet2Dial, here are the steps to add the SMS functionality. Just log into your server as root and issue the following commands. For Mac users, there is no root account. Just open a Terminal window while logged in with the user account used to set up Tweet2Dial initially and skip the cd /root command below:
cd /root
mv tweet2dial.php tweet2dial2.php
wget http://pbxinaflash.net/source/twitter/tweet2dial.tgz
tar zxvf tweet2dial.tgz
rm tweet2dial.tgz
Now open your old Tweet2Dial application (renamed to tweet2dial2.php) and write down your existing settings. Then edit tweet2dial.php and plug your old settings back in to restore access to your Google Voice account, your Asterisk server (if desired), and your Twitter friends. That's it! You're finished.
Sending SMS Messages with Twitter. To send new SMS messages, you'll use the same scenario outlined in the original article to place free phone calls. Just send a direct message to your secondary Twitter account. Only those that you have authorized as friends can send direct messages to this account so it's as secure as you want it to be. The Twitter Direct Message syntax for an SMS message looks like this where 6781234567 is the 10-digit cellphone number or Google Voice number of the SMS recipient:
SMS:6781234567:Here is a sample SMS message
Any replies to an SMS message which you send using Twitter will be forwarded to the email address that you used to set up your Google Voice account. Enjoy!
Special Thanks. Our tip of the hat again goes to the Pygooglevoice Development Team: JEIhrig, justquick, jacob.feisley, and nagle. Without their pioneering work, there would be no Tweet2Dial, no Orgasmatron V, and no Googlified Messaging for Asterisk. Terrific code! Thank you.
Happy Birthday to Us! Well, today's the Big Day. Today marks the Fifth Birthday for Nerd Vittles. Seems like only yesterday. Thanks for putting up with us all these years!
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.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
Tweet2Dial: Free Google Voice Calling & SMS with Twitter
To celebrate the New Year, it seemed only fitting to bring Google Voice calling out of the cloud and into our favorite social hangout. For our special New Year's project, we're pleased to introduce Tweet2Dial. It lets you use Twitter or your favorite Twitter client to make free outbound calls through Google Voice to anyone in the United States or Canada. Just send a Direct Message to your new Twitter account and, in less than a minute, your phone will ring connecting you to the person's phone number you specified in your Twitter message. In addition, you also can send SMS messages to anyone with an SMS-capable device in the U.S. and Canada. All of this magic is managed on your existing Asterisk® server or almost any Linux server or Mac. There's no Asterisk overhead to process the calls and SMS messages because Asterisk isn't required! But, to start 2010 off on the right foot, we've included a little bonus at the end of this article for all the Asterisk administrators in the house. If you happen to be using an Asterisk server, you now can manage it from Twitter with Tweet2Dial, too.
For those with cellphone plans that let you designate certain numbers for free, unlimited calling (such as Sprint, AT&T, Verizon, and T-Mobile), adding your Google Voice number to your preferred number list will mean that all of your Tweet2Dial-originated cellphone calls to anyone and everyone throughout the U.S. and Canada will now also be totally free with no impact on your bucket of call minutes.
Yes, we know Jajah is working on something similar for Twitter. But you have to be invited to participate in Jajah's beta (we didn't make the cut!), free calls are limited to two minutes, and both parties have to have a Twitter account which doesn't work too well for calling grandma. So why put up with all the limitations and restrictions of Jajah when you can do it yourself?
There's been some tech chatter that the procedure we've outlined below is complicated. If you can paint by number or bake cookies from the back of a Nestle's bag, trust me. You can handle this! Getting a Mac or a Linux server set up to support Tweet2Dial only takes a minute or two. So ignore the trade rags. Some of them can barely read. 🙂
If you've already gone through our Google Voice tutorial which enables free Google Voice calling on your Asterisk server, or if you've installed our all-in-one Orgasmatron V build on your Asterisk server, or if you have a Mac or you've built your own Linux server without Asterisk, there's no need to wait for Jajah and no need to limit your calls to two minutes or to those with Twitter accounts! You can call anyone in the United States or Canada right now, talk as long as you like, and do it all for free with Tweet2Dial, Twitter, and Google Voice! If you're a Windows user, check out the Google Voice Dialer for Windows.
Prerequisites. To get started, you can use your Asterisk server configured for Google Voice as we've outlined above. We won't actually be using Asterisk to place the calls, but our previous tutorials get your server properly set up with Google Voice and the latest, awesomest1 pygooglevoice to support Tweet2Dial. Any of the Asterisk aggregations such as PBX in a Flash will work great.
If you don't have a PBX in a Flash server with Google Voice already configured, shame on you! Just kidding. Actually, any recent CentOS or Fedora Linux server will work just as well today. Log into your server as root. Run rpm -q python to make sure you have at least Python 2.4 installed on your system. If not, run: yum update python. Then execute the following commands:
cd /root
yum install python-setuptools
easy_install simplejson
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice*
cd pygooglevoice-0.5
python setup.py install
Tweet2Dial also will run just fine on any Mac of recent vintage. We've actually tested it with Snow Leopard. Basically, to get Python and Apache set up properly, you have to enable root access, switch to root user access with su in Terminal, activate PHP support in Apache, turn on Web Sharing in System Preferences->Sharing, run easy_install simplejson as root to install simplejson (the Python Setup Tools already are in place!), using a browser download pygooglevoice to your Downloads folder, untar it as root in Terminal with the same command as above, and then while still logged into Terminal as root, go to the Downloads/pygooglevoice-0.5 folder and run the following command: python setup.py install. The only variations in the Tweet2Dial setup will be the storage location for Tweet2Dial (there is no root folder on a Mac) and the methodology for setting up the crontab entry (HINT: we'll run crontab -e to add a crontab entry since there is no /etc/crontab file). Just follow along using the Mac-specific instructions below for details, and everything will work swimmingly.
To test whether your server is properly configured for Tweet2Dial, log in as root and type: gvoice. You should be prompted for an email address. If so, press Ctrl-C to exit. You're ready to roll. If not, pygooglevoice has not been properly installed on your server.
You'll obviously need a Google Voice account. Request an invite here or just post a brilliant comment below, and one might magically appear in your inbox. Configure your Google Voice account with all the phone numbers from which you want to place outbound calls. One of these numbers will already be the go-between number for Google Voice and your PBX in a Flash server (IPkall or SIPgate) if you've followed our previous tutorials. Now simply add additional numbers that you want to use to place outbound Google Voice calls. This would include numbers such as your cellphone, your vacation home, and your direct-dial office number. You do not need to enable them for ringing when inbound calls arrive on your GV number.
For today's project, you'll also need a new Twitter account even if you already have one. Why? Because you can't send a Direct Message to yourself with Twitter. So we'll use your primary Twitter account to send Direct Messages with dialing instructions to your secondary Twitter account. Then we'll use Tweet2Dial to poll your secondary account and retrieve the dialing instructions to actually place the outbound calls with pygooglevoice through your server. It sounds harder than it actually is. Honest! Assuming you already have Google Voice running on your Asterisk server, you'll be tweeting away in 10 minutes. If you have a current Linux server, add an extra 2 minutes to install pygooglevoice using the steps above.
Usage Considerations. Before someone asks, let's address Question #1. Can others send messages to my Twitter account in order to make outbound calls through my server using Google Voice? And the answer is yes and no. We're going to configure your new secondary Twitter account with Protect My Tweets enabled. This means you have to approve friends and also become their friend before they could send a Direct Message to your secondary Twitter account. So, yes, if you approve, any Twitter user could theoretically place calls using your Twitter secondary account. For the average reader, we wouldn't recommend it for a couple of reasons. Here's why.
Google Voice only lets you link a handful of phone numbers to your GV account. So, for your friends to be able to place calls using your GV credentials, you'd have to forfeit one of your allotted quota of numbers for each person... or their phone would never ring to place the outbound calls. Yours unfortunately would! Remember, Google Voice always places two calls to complete a connection: one to you (using one of the phone numbers defined in your GV account) and one to the person with whom you wish to speak.
The other reason for not opening this up to other callers is that Google Voice limits your account to one outbound call at a time. If others are using Twitter to make calls using your GV credentials, it means you can't. And there's no mechanism for easily identifying when a call already is in progress. So our recommendation is to keep your secondary Twitter account private and set up Following and Follower linkage only with your primary Twitter account. This will mean that Direct Messages to your secondary Twitter account can only originate from your primary Twitter account. You can still place outbound calls to anybody, but others can't!
Having said all of that, we've designed Tweet2Dial so that you can allow others to use your secondary Twitter account to place Google Voice calls using their own GV credentials. This saves them the aggravation of setting all of this up, but it means they have to trust you enough to share their Google Voice credentials. After all, what are friends for? 😉 At the end of this article, we'll walk you through how to do this if you really have the urge. We would hasten to add that the actual processing load on your server is virtually zero so don't be deterred by performance concerns. Pygooglevoice sends the calling instructions to Google Voice, and then your server is completely out of the call loop. We've still limited outbound call setup to one call per minute, but these calls do not have any impact on Asterisk resources and only very minimal impact on your server. The only drawback to hosting Tweet2Dial for your friends is that, if five simultaneous Twitter messages are sitting in the queue, it would mean the last call request won't be processed until about 5 minutes after the Twitter message was sent. But, unless you have a bunch of extremely chatty friends, call request congestion shouldn't be a problem.
One final word of caution. Twitter currently permits a maximum of 150 Twitter API calls per hour per account. There is some good news. Within the next few weeks, this limit will be increased to 1500 per hour, but it hasn't happened yet. This application is designed to poll your secondary Twitter account once a minute to retrieve and then discard your oldest, existing Direct Message. So it uses 120 of your allotted 150 API calls per hour to work its magic. You are well advised NOT to run any third-party Twitter applications with this secondary Twitter account, or you will quickly exceed the current connection limitation. When the API limit is reached, it means none of your pending call requests would be processed until the next hour rolls around... at least until Twitter raises this connection limit. Once Twitter raises the API limit, we may revisit our code and eliminate the current one call per minute limitation. So stay tuned!
Creating A Secondary Twitter Account. First, let's get your secondary Twitter account set up. Go to twitter.com and create a new account with a very secure password! You must enter a different email address than the one used for your primary account. Use one you can actually access! Log into your new account and choose Settings. Scroll down to Protect my tweets and check the box by clicking on it. Save your settings. NOTE: This check box is critically important. It keeps the entire world from being able to access your server! There are other layers in the security model, but this one is VERY IMPORTANT so verify it twice! Now log back into your primary account. Then goto http://twitter.com/SecondaryAccountName and request access. You'll get a message that your request for access has been sent. Log out and back into your secondary account once again. Authorize your primary account name as a Follower. Now log out and back into your Primary Account. We'll use it to send a Direct Message to your secondary account in a few minutes.
Installation and Configuration. To install Tweet2Dial, log into your server as root and issue the following commands:
cd /root
wget http://pbxinaflash.net/source/twitter/tweet2dial.tgz
tar zxvf tweet2dial.tgz
rm tweet2dial.tgz
If you're doing this on a Mac, there is no wget application and no root folder so you'll need to download tweet2dial.tgz with your browser. Save it to your Downloads folder. Then open a Terminal window and execute this command:
tar zxvf Downloads/tweet2dial.tgz
Now let's configure the application:
nano -w tweet2dial.php
At the top of the file, you'll see the following lines:
// Your SECONDARY Twitter account username and password
$username = "TwitterUsername";
$password = "TwitterPassword";// Authorized Twitter users with corresponding GV credentials go below
$user['twitname'][1]="YourPrimaryTwitterUsername";
$user['gvemail'][1]="YourGoogleVoiceEmailAddress@gmail.com";
$user['gvpass'][1]="YourGoogleVoicePassword";
$user['gvcall'][1]="6781234567";// *** Leave everything below this line alone. 🙂
Begin by entering your secondary Twitter name and password by replacing TwitterUsername and TwitterPassword with your actual credentials. Be careful here. Capitalization matters! If you set up your Twitter username as gvNerdUno, don't enter gvnerduno! Now move down to the four $user entries. The first is your primary Twitter account name. Replace YourPrimaryTwitterUsername with your actual Twitter account name. Again be careful of capitalization! Next, enter the login email address for your Google Voice account replacing YourGoogleVoiceEmailAddress@gmail.com. Next, enter your Google Voice password replacing YourGoogleVoicePassword. Finally, enter one of the 10-digit ringback numbers you've configured in your Google Voice account by replacing 6781234567. Do NOT use the one that's reserved for use by Asterisk! This is the number that will be called by default whenever you place an outbound call with Twitter. You'll have the option of overriding it, but this saves your having to enter both a destination phone number and a callback number each time you wish to place a call. Be sure to preserve the quotes around each of the entries. Once you've double-checked all of your entries for typos, save your changes: Ctrl-X, Y, then Enter.
Tweet2Dial Test Drive. Now that everything is set up, let's place a test call to be sure everything is working. Log into your primary Twitter account. Click on Direct Messages. Choose your secondary Twitter account from the pulldown menu. In the block below Send a Direct Message, enter a 10-digit number in the U.S. or Canada that's different from your default callback number. Then click the Send button. It's that simple! Once Twitter tells you the message has been sent, log into your Asterisk server and execute the following commands.
cd /root
./tweet2dial.php
If you're on a Mac, just open a Terminal window and type ./tweet2dial.php. In either case, you should get a response indicating that your call has been placed, and your default phone number should begin to ring. When you answer it, Google Voice will place a call to the 10-digit number that you entered in your Twitter direct message above.
Now, just for fun, run Tweet2Dial again: ./tweet2dial.php. If everything is working properly, you will see the following message: Nothing to do.
Finally, assuming you have configured another callback number in Google Voice that is close at hand and not your Asterisk callback number, send another Twitter direct message with the following syntax: 8439876543:6781234567 where 8439876543 is the 10-digit number of someone you wish to call and 6781234567 is a 10-digit ringback number already set up in your Google Voice account. Once the message has been sent, run Tweet2Dial again from the command prompt.
When you're sure everything is working reliably, add the following entry to the bottom of /etc/crontab unless you're using a Mac. This will run the application once a minute around the clock looking for incoming Twitter messages:
* * * * * root /root/tweet2dial.php > /dev/null
If you're running this on a Mac, add an entry to your crontab like this. From the Terminal window, run: crontab -e. Once the vi editor opens, type:
* * * * * /users/youracct/tweet2dial.php
Substitute the name of your Mac account for youracct. Then press the Esc key followed by :wq. Check your work by typing: crontab -l. Your entry should look like this:
* * * * * /users/youracct/tweet2dial.php
Sending SMS Messages with Twitter. To send SMS messages using Twitter, you'll use the same scenario outlined above to place free phone calls. Just send a direct message to your secondary Twitter account. Only those that you have authorized as friends can send direct messages to this account so it's as secure as you want it to be. The syntax for an SMS message looks like this where 6781234567 is the cellphone or Google Voice number of the SMS recipient:
SMS:6781234567:Here is a sample SMS message
Any replies to an SMS message which you send using Twitter will be forwarded to the email address that you used to set up your Google Voice account.
For Whiz Kids Only. Now let's say you want to let your spouse use her Twitter account to place calls using her very own Google Voice credentials. First, you need to authorize her as a follower in your secondary Twitter Account. Second, you need to add a new block of code in tweet2dial.php that looks like the following. Place it immediately below the existing $user entries in the file:
$user['twitname'][2]="SpousePrimaryTwitterUsername";
$user['gvemail'][2]="SpouseGoogleVoiceEmailAddress@gmail.com";
$user['gvpass'][2]="SpouseGoogleVoicePassword";
$user['gvcall'][2]="6781234567";// *** Leave everything below this line alone. 🙂
Notice that the only change is this array subset is numbered [2] while the original was numbered [1]. You can add as many as you like so long as you increment this number and provide the credentials for each user. Now you have your own little Jajah-like sandbox, and it's absolutely free.
For Asterisk Administrators Only. Want to manage your Asterisk server from Twitter? There's an app for that. We promised you a New Year's bonus so here it is. First, read our last article which explains how to manage your Asterisk server using email messages and the Asterisk CLI. Now you can do exactly the same thing using Twitter direct messages. The only Twitter user that can do this on your server is the Twitter account name you specified in the #1 $user slot above. So you don't have to worry about your pals trashing your Asterisk server if you give them privileges with Tweet2Dial. The syntax for issuing CLI commands using Tweet2Dial looks like this:
CLI: database show cidname 8437978000
Just be sure Direct Messages from your primary Twitter account begin with CLI in all CAPS followed by a colon, a space, and then the desired CLI command. That's all there is to it. You'll get a confirmation Direct Message in your main Twitter account once the command has been executed assuming you have established Following and Follower linkage between your primary and secondary Twitter accounts. Test sending DMs in both directions to double-check it. And if you've enabled email delivery for Direct Messages in your Twitter configuration, you'll get an email confirmation as well. Because of Twitter's 140 character limitation, some commands such as help don't provide all of the output you normally would receive from the CLI. You'll only get the last line. Aside from that, the CLI functionality is identical to interacting directly with the Asterisk CLI and the email implementation we outlined previously. Here's the CLI response:
Before you can use the CLI interface in Tweet2Dial, you have to enable it. Edit tweet2dial.php and change $CLIenable=false to $CLIenable=true. And, yes, we understand there are some of you that don't trust Twitter to keep your commands secure. Well, first of all, in order to penetrate your Asterisk server, someone would have to send a Twitter Direct Message from your primary Twitter account. So they'd need your password and they'd need to know the syntax for Asterisk CLI commands AND the syntax for sending them via Twitter. But, there's always a Cracker Rapper2 somewhere. Right? So we've also built a password into the system at your server's end so you can sleep more comfortably. The default password is CLI. But feel free to change it to anything you like. Just edit tweet2dial.php and find this line: $CLIpword = "CLI";. Replace CLI (between the quotes only!) with whatever password you'd like. After saving your changes, you'll need to adjust your Twitter messages accordingly. For example, if you changed your password to FooBar, then your future Twitter CLI command syntax would look like this: FooBar: help. Enjoy!
Special Thanks. As Nerd Vittles prepares to celebrate its Fifth Birthday, we want to take a moment to thank those that have made Nerd Vittles and the PBX in a Flash project possible. Without the generous financial support of Vitelity and Google's AdSense program plus the unwavering support of our hosting providers who provide free downloads of PBX in a Flash around the globe, all of what we do would be much more difficult and expensive! It's not too late for you to kick in a nickel or two as well if a fleeting moment of generosity should strike. 😉 There's a Donate button at the top of the page. Finally, we want to thank Digium® for their continuing support of the Asterisk project and their generous contribution of hardware to the PBX in a Flash development team during 2009. Happy New Year everybody!
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.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
Free U.S. & Canada Calls: Google Voice Dialer for Windows
There now are a number of ways to make free calls to anyone in the U.S. and Canada using Google Voice without having to jump through the hoops of calling into your voicemail and having Google Voice call you back. There’s our Asterisk® implementation using pygooglevoice which lets you transparently place calls through Google Voice using any phone connected to your PBX in a Flash system. You also can set up a Sip Sorcery account and make free calls through that interface using a SIP phone. And now there’s Dogface05’s stand-alone Dialer for Windows that lets you place calls from the Windows command line in seconds. Because this is such a simple alternative, everyone should add it to their Windows toolkit. Here’s how.
Prerequisites. You’ll obviously need a Google Voice account. If you don’t have one, just register for an invite. Next, you’ll need a phone number to use for placing the outbound calls. And, finally, you’ll need to download and install Dogface05’s dialer on your Windows system.
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of the phone you’ll be using to place calls and mark it as an Office phone. You’ll have to go through Google’s confirmation drill to successfully register the number with Google Voice. After the number is confirmed, be sure there’s a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to this number.
While you’re still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller’s Number. Finally, uncheck Do Not Disturb. Now click the Save Changes button.
Dialer Setup for Windows. From your Windows machine, open a browser and download the Google Voice dialer to your Desktop. Unzip the downloaded file and drag gvdial.exe to your \windows directory so that it’s in your path.
Placing a Call. Let’s first make sure everything is working properly. Open a command prompt window from the Windows Desktop and enter a dialing command using the following syntax:
gvdial username password destination ani [phonetype]
where:
- username = your Google Voice email address
- password = your Google Voice password
- destination = 10-digit number of person to call
- ani = your 10-digit phone number registered with Google Voice
- phonetype = 3
The phonetype is actually optional and can be ignored unless you happen to be using a Gizmo number in which case it needs to be 7. Never enter the brackets. That merely signifies that the entry is optional.
Assuming your registered email address with Google Voice is joe@gmail.com, your password is secret, the number you wish to call is 6781234567, and your number is 4049876543, the dial string should look like this:
gvdial joe@gmail.com secret 6781234567 4049876543
Your phone should ring at this point, and Google Voice will complete the outbound call to 678-123-4567.
Creating Speed Dial Batch Files. Using Notepad, you now can create batch files for frequently dialed numbers. For example, the entry above could be saved in a batch file called joe.bat. Then simply create a desktop icon for Joe and link it to joe.bat. Double-click on the Joe icon whenever you wish to place a call to Joe. Here’s how the batch file might look:
echo off
cls
gvdial joe@gmail.com secret 6781234567 4049876543
echo Press ENTER key after the called party answers.
pause
Surfing the Google Wave. We’ve got a dozen Google Wave invites to give away during the next week. Just post a comment on any Nerd Vittles article, and we’ll put your name in the hat. Be sure to provide a Gmail address with your comment as this is required to take advantage of the Google Wave Preview. Here’s a sample for you to try once you have Google Wave credentials:
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.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Tweaking Asterisk for Free Google Voice Calling
Now that the Asterisk® and Google Voice marriage is finally underway, we wanted to step back today and revise the original methodology a bit to take advantage of some of the terrific comments which were offered in response to our last article. First, the good news. U.S. calls through Google Voice using Asterisk work! They sound great, and they're free. The not so good news was that the MeetMe conferencing trick to join your outbound call with the Google Voice click-to-dial return call from your destination worked great so long as a real person answered the phone. But, if an answering machine picked up or no one answered the call at all, there were problems because these calls already had been transferred to the MeetMe conference and there was no simple way to disconnect them. And the need for two DIDs to support a single Google Voice interface just seemed a bit wasteful.
9/1/2010 Update: A good bit has changed with Google Voice since this article was first published. For the definitive guide and installation procedure, we highly recommend The Incredible PBX and accompanying article which can be found at this link. Google Voice (and much more) already is included in our new PBX which is literally Plug-and-Play. If you prefer to roll your own, be sure to also have a look at this excellent update on the Michigan Telephone Blog.
Today we want to try to eliminate these two quirks while stiill providing a seamless interface between Google Voice and Asterisk. We also appreciate that thousands of you already have implemented the previous approach. So we want your transition to the new way of doing things to be as painless as possible. On the other hand, for frequent readers, we hope you'll bear with us as we repeat some of what already has been covered in previous articles so new visitors don't have to jump around between articles to get the complete picture of what we're trying to accomplish.
The objective remains the same. We want a methodology that lets us make outbound calls from any Asterisk phone using the Google Voice service to take advantage of free calling in the United States and Canada. And we want calls to our Google Voice number delivered to our Asterisk system for transparent call processing. Yes, SIP is still on our wish list for both outbound and inbound calls with Google Voice, but we'll make do with PSTN calls particularly while Google is footing the bill for all of the calls.
Update: There's now a turnkey Asterisk solution that implements Google Voice calling without getting your hands dirty. Check out our new Orgasmatron V.
Tweaked Design. Here's the new design. You obviously still need a free Google Voice account. If you don't have one, you can request an invite here. At last report, it's only taking a few days from application to invite which is really great news. Don't use a space in your Google Voice password! Once you have a Google Voice account and phone number (Google has reserved a million of them so... not to worry!), then you'll need a DID that provides unlimited, free incoming calls. Once you get your DID set up on your Asterisk system, we'll set up a forwarding phone number for this DID in your Google Voice account so that Google Voice calls can be connected to your Asterisk server.
For outbound calls, we'll combine a little dialplan voodoo with pygooglevoice to instruct Asterisk to place a click-to-dial call using your Google Voice forwarding number. Then we'll stuff in the destination U.S. phone number. When you dial GV-678-1234567 from any of your Asterisk phones, Asterisk will park your initial call in a reserved parking lot slot and then join the called party to the originally parked call. The entire procedure is virtually transparent both to the caller and the callee. And, unlike the MeetMe conference, the parking lot fades out of the picture as soon as the call is connected. Thus, if either party hangs up, the active channel for the call is terminated on your Asterisk server.
For inbound calls from your Google Voice number, we'll tweak the dialplan so that it can distinguish between a RingBack call that Google Voice initiated and a true inbound call. We'll peel off the real inbound calls and route them to a separate Inbound Route in FreePBX for processing in any way you desire.
Finally, for those that implemented the methodology in our previous article, we'll walk you through the steps to revise your existing setup to take advantage of these new tweaks. You can skip over the initial installation process if you already have gone through the Google Voice setup from our earlier article. Just skip down to Tweaking Previous Setups.
Special Thanks. At the outset, we again want to express our sincere appreciation to Jacob Feisley and Paul Marks for their pioneering work on a Python interface to Google Voice. We also stumbled upon another Python development project, Google Voice for Python. While we originally had planned to rely upon Jacob and Paul's script, we ultimately decided to implement pygooglevoice because of the additional flexibility it provided for down the road. With pygooglevoice, you not only can make Google Voice calls, but you also can send SMS messages with no muss or fuss. Jacob Feisley has now joined that project as well. So, our special tip of the hat goes to the entire Google Voice for Python development team. It's a terrific product as you will see.
Prerequisites. Today's setup requires a CentOS-based Asterisk aggregation with a current version of FreePBX. Be aware that today's solution requires Python 2.4 or higher and reportedly will not work with Python 2.3 found in some Linux distributions. We've tested everything with PBX in a Flash and, on that platform, you're good to go. The install script should work equally well with the other CentOS-based Asterisk aggregations, but we haven't tested them. Be our guest, and let us know if you encounter any problems. Finally, a word of caution. We don't ordinarily distribute solutions using development tools we don't use. Our knowledge of Python wouldn't fill a thimble. We've made an exception today because of the extraordinary interest in Google Voice by the Asterisk community. But, if something comes unglued, we can't fix it. So have a backup plan in place just in case. 🙂
Today's Drill. To get everything working today, there are six steps: (1) obtaining and configuring a DID to manage calls between Google Voice and Asterisk, (2) configuring a Google Voice forwarding number for this DID to manage your outbound and inbound calls, (3) configuring FreePBX to route all outbound calls with a GV prefix to your special Google Voice dialplan context, (4) configuring an inbound route to manage incoming calls from your Google Voice number, (5) setting up a series of Parked Call extensions, one of which will be used to manage your outbound Google Voice calls, and (6) running our install script which adds the dialplan code for Google Voice calling with your credentials and puts the Python application into place on your server. It sounds more complicated than it is. So hang on to your hat. Here we go!
Dedicated DID. Before you can use Google Voice with Asterisk, you'll need a DID that can be dedicated to your Google Voice interface to Asterisk. We'd recommend a free IPkall or SIPgate DID. To get started, use one of the links above to obtain and configure the DID. Temporarily point the DID to an extension on your Asterisk system that can be used to verify your requests for the number. Since all of these calls are free, the area code of the DID really doesn't matter because you're never going to publish the fact that it exists.
The easiest method for setting up the DID is to first create a SIP URI for the DID on your Asterisk system. Next route the SIP URI to an Inbound Route in FreePBX where you can manage the destination for calls to that DID. Initially, you want the destination to be an extension on your Asterisk system that you can answer to verify both the DID setup and the GV setup below. Finally, point the DID you obtained to the SIP URI defined above.
HINT: The entry in extensions_override_freepbx.conf would look something like this for a SIP URI called ipkall-1:
exten => ipkall-1,1,Goto(from-trunk,${DID},1)
Then you would create an inbound route named ipkall-1 using FreePBX and designate some existing extension on your server as the destination for these inbound calls.
When you set up the SIP forwarding for the DID at ipkall.com, you'd specify the SIP URI as:
ipkall-1@ipaddress_of_your-Asterisk_server
We've previously covered in detail how to do this so read the article if you need a refresher course. To reiterate, the area code of this DID really doesn't matter because you're never going to give out the number. So use one of the free sources and save yourself some money. The real trick is you want to use a DID with unlimited, free inbound calls. Both IPkall and SIPgate provide that functionality at no cost.
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of your DID. Be sure the DID is pointed to an extension on your PBX that you can answer since you have to go through Google's confirmation drill to successfully register the number. After the DID is confirmed, be sure there's a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to your Asterisk server.
While you're still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller's Number. Be aware that IPkall DIDs only forward your IPkall number as the CallerID number while SIPgate DIDs reportedly forward the actual number of the person calling you. If this matters to you, then you may prefer the SIPgate DID option. Finally, uncheck Do Not Disturb. Now click the Save Changes button.
Integrating Google Voice into Asterisk with FreePBX. Open FreePBX with a web browser and choose Setup, Trunks, Add Custom Trunk. Insert your GV number in the Outbound CallerID field and add the following Custom Dial String on the form and Submit Changes and reload the dialplan:
local/$OUTNUM$@custom-gv
Next, 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-gv
Inbound Routes. Next, we need two Inbound Routes to get everything working. In setting up your DID with IPkall or SIPgate, you already should have created one inbound route for that provider. It already should be routing calls to an extension on your PBX. Now we need to create a Custom Destination for this inbound route and then reroute these calls there. In that way, your RingBack calls will be routed to some special dialplan code that drops these calls into a custom parking lot where the RingBack call is married up to the extension from which you placed the original call. Then we need to create another inbound route to manage normal incoming calls that are forwarded to your PBX whenever someone dials your Google Voice number.
To begin, choose Tools, Custom Destinations, Add Custom Destination and add an entry like this and then click the Submit Changes button:
Custom Destination: custom-park,s,1
Description: Custom GV-Park
Next choose Setup, Inbound Route and click on the inbound route you created previously for IPkall or SIPgate. Change the destination for these calls to Custom Destination: Custom GV-Park.
Now click on Add Incoming Route and create a new route for your incoming Google Voice calls. Give it any description you like but, for the DID number, it must be gv-incoming. You can leave most of the other defaults. Just be sure you set a destination for your incoming calls from Google Voice. It could be an extension, ring group, IVR, or whatever best meets your needs. The important entry here is gv-incoming for the DID number. Click the Submit button to save your entries. Ignore the warning that you've entered an oddball DID. We know what we're doing. 🙂
Setting Up the Parking Lot. While still in FreePBX, we need to create or adjust your existing settings in Setup, Parking Lot. The parking lot is used by FreePBX to simulate old key telephones where you could place a call on hold and then someone else in the office could pick up the call by clicking on the blinking key on their phone. The Asterisk equivalent is to press the flash hook and dial your Parking Lot Extension which then places the call in a Parking Lot space and tells you what the space number is. Someone else then can dial the number of that space to pick up the call. Our little trick today works like this. When you place an outbound call through Google Voice, your extension will be dumped into a reserved parking lot space. When Google Voice initiates the RingBack call before connecting the destination number you've dialed, that call will be sent to the same reserved parking lot space. The two calls then are joined, and you'll hear the parking lot number followed by ring tones as your call is connected by GV to its final destination. Our special thanks to Richard Bateman for his comment on the previous article and this terrific tip! He wins an Atomic Flash installer from Nerd Vittles. In addition, A. Godong wins an Atomic Flash installer for his tip on consolidating two DIDs into a single DID to manage both inbound and outbound GV calls. Just send us your addresses.
Now, where were we? Most FreePBX systems have a default setup for the Parking Lot. What we need to do is be sure you have reserved one more space in the parking lot than you actually need for day to day operation of your PBX. We'll use the last parking lot space number to manage outbound calling through Google Voice. Our entries look like the following:
Enable Parking Lot Feature: checked
Parking Lot Extension: 70
Number of Slots: 5
Parking Timeout: 30 seconds
Parking Lot Context: parkedcallsDestination for Orphaned Calls: Terminate Call: Hangup
If you use our setup above, the Magic Number is 75 which is the fifth slot in the Parking Lot. If you use a different Parking Lot extension or number of slots, here's how to calculate the Magic Number. Start counting the slots beginning with one more than the Parking Lot Extension. When you get to the last slot in the number of slots you've specified, that's your Parking Lot Magic Number. Write it down. You'll need it in a second when you run our GV installation script.
Save your entries and reload the Asterisk dialplan when prompted.
Integrating pygooglevoice. Now we're ready to complete the setup by running our revised script which loads pygooglevoice and sets up your dialplan in extensions_custom.conf. You'll need 5 pieces of information to run the script so write them down before you begin:
1. Your 10-digit Google Voice phone number
2. Your Google Voice email address
3. Your Google Voice password (no spaces!)
4. Your 11-digit RingBack DID (16781234567)
5. Your Parking Lot Magic Number
A word of caution: If you used a gMail address to set up your Google Voice account, it's possible to have different gMail and Google Voice passwords. For this to work, you'll need to enter your gMail password, not your Google Voice password (assuming they're different).
Now log into your Asterisk server as root and issue the following commands:
cd /root
wget http://bestof.nerdvittles.com/applications/gv/install-gv-new
chmod +x install-gv-new
./install-gv-new
Google Voice Speed Dials. For frequently called numbers, you can add speed dials by inserting entries in the [from-internal-custom] context of extensions_custom.conf that look like the example below where 333 is the speed dial number and 6781234567 is the area code and number to call. Be sure to reload your Asterisk dialplan to activate them.
exten => 333,1,Dial(local/6781234567@custom-gv,300)
Congratulations! You now have what we hope will be flawless and free U.S. calling on your Asterisk system using Google Voice. No gimmicks, no strings, no cost. Enjoy!
Finally, one additional word of caution. Both Google Voice and this call design are set up for a single call at a time. There are no safeguards to prevent multiple calls, but that may violate the Google Voice terms of service.
Asterisk 1.6 Solution. Several readers now have documented the procedure for implementing the Asterisk 1.6 bridge technology to make outbound Google Voice calls. You can read all about it here.
Tweaking Previous Setups. If you installed pygooglevoice using our previous tutorial, here's what you need to do. First, log into your Asterisk server as root and issue the following commands:
cd /etc/asterisk
nano -w extensions_custom.conf
Scroll to the bottom of the file by pressing Ctrl-W then Ctrl-V. Move up the file using up arrow until you reach [custom-gv]. Press Ctrl-K repeatedly to delete all of the lines in the [custom-gv] context. If you get to another line that starts with a label in brackets like [this], STOP deleting. Once you've deleted all of the lines in the [custom-gv] context, save the file: Ctrl-X, Y, and press Enter.
Now continue reading this article by jumping up to the Google Voice Setup topic. The Custom Trunk entry and the GoogleVoice outbound route will already be in your FreePBX system so there's no need to repeat those two steps. You will need to perform the remaining FreePBX steps beginning at the Inbound Routes topic and continuing on with Setting Up the Parking Lot. Finally, when you run the new installation script, it will detect that pygooglevoice is already on your system and will skip that step but will install the new custom contexts in extensions_custom.conf using your new settings. Enjoy!
Thought for the Day. Which is more arbitrary: (1) Apple snubs Google Voice or (2) Google Voice snubs SIP? Pays to look in the mirror occasionally.
Best Read of the Week. Memo to Steve Jobs and Apple: Stop Being A Jerk!
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.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
Nerd Nirvana: Free Google Voice Calling Returns to Asterisk
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...
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.
Originally published: July 26, 2009
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Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
Googlified Messaging Returns: The Gizmo-Asterisk Marriage
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!
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
- Backdoor Dialing unfortunately has bitten the dust at least temporarily. [↩]
Googlified Messaging: Asterisk’s New Best Friend
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 Dialing and ENUM interfaces that didn't cost us a dime. And, today, we turn our attention to Google's recent transmogrification of GrandCentral into Google Voice. More specifically, what we want to do is examine some ways to integrate the Google Voice feature set into our existing Asterisk implementations. The potential benefits are enormous. There's free calling in the U.S., free distribution of inbound calls to multiple phone numbers scattered around the country, free SMS messaging and delivery by email, free transcription of voicemail messages into text-based emails, free conferencing, and free GOOG-411, a voice-activated service that let's you find nearby businesses by saying where you are and what you're looking for. For today, we've set our sights on the Google Voice feature set which is easiest to integrate into existing Asterisk systems: free voicemail message transcription, free calling in the United States, and free GOOG-411 directory assistance. For lack of a better term, we call it... Googlified Messaging™. 😉
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.
Integrating Google Voice into Asterisk. If there is a recurring theme to Google Voice, it's this. Google Voice was designed to be a user-friendly, interactive messaging system. Google didn't intend to provide a telephony toolkit for Asterisk developers, but they haven't blocked any functionality either. There's no SIP connectivity in Google Voice... at least that is obvious. Can you spell G-I-Z-M-O? Well, that was the first hint. But a simple call trace revealed a lot more. It appears the entire Google Voice platform is SIP-based which makes it a perfect fit with Asterisk.
Because of the Google Voice design, there's no simple way to use your Google Voice DID for incoming call distribution while also integrating voicemail transcription and outbound calling into your Asterisk dialplan. Why? Because you can't take advantage of the free voicemail transcription service with Asterisk if Google Voice is sending inbound calls all over the countryside. So the real key to unlocking the greatness of Googlified Messaging is having two Google Voice accounts so that each can be used for a dedicated purpose. The first account will be used for outbound functions and voicemail transcription while the second is used to manage and route incoming calls. This is important because, for security reasons, you don't want to reveal your Google Voice number that is being used for outbound calling. Why? Because it is a SIP connection, and your Google Voice phone bill is only protected by a 4-digit PIN. If Google hasn't learned about Fail2Ban, they will soon. As this is written, multiple Google Voice accounts aren't possible unless you had more than one GrandCentral account since only GrandCentral users currently are eligible for Google Voice accounts. But that, too, will change!
For today, let's put aside the incoming call routing and concentrate on the remaining Googlified Messaging functionality. We turn first to Google Voice's free transcription of voicemail messages into text-based messages for email delivery to your desktop PC or cellphone.
Voicemail Transcription Overview. We begin with a cautionary note. Google's new automated voicemail transcription service is absolutely incredible... even if it's not quite perfect. We've tried a couple of messages to evaluate the transcription accuracy, and we'll let you judge for yourself.
Actual Message: "Hi. I was just passing through the airport. I hadn't seen you in a couple years, and I thought you might wanna get together for a quickie. Give me a call."
Googlified Transcription: "hi i was just passing through the airport i hadn't seen you in a couple years and i thought you might wanna get together for a quickie give me a call"
As you can see, the accuracy was pretty good. But there are a couple of problems. First, there's no CallerID name associated with inbound calls. So, if the caller doesn't identify himself or herself (especially if the caller is using a pay phone), you're S.O.L. relying on the transcription. But the message and phone number were accurate. It probably would motivate you to quickly connect to your email account and actually listen to the voicemail to decipher the caller's identity and avoid a missed opportunity. 🙂
Actual Message: "Hi. I've read over your corporate acquisitions and merger paper, and it isn't quite accurate with regard to our position."
Googlified Transcription: "hi i have a red over your corporate acquisitions in merger paper and it is a quite accurate with regard to our position"
This second example is a bit more problematic. The same issues apply from the first example. Plus there's a new wrinkle that could be a show stopper: the Googlification of "isn't quite accurate" into "it is a quite accurate." You'd better hope there was more to the message than this before running off to present your paper. It also highlights the difficulty that automated systems have when deciphering conjunctions such as "isn't" which often are used in conversational speech.
Some might suggest that this demonstrates the Google developers actually have their priorities in order. Get the kinks out of the sex jargon before focusing on exciting subject matter such as conjunctions. 🙄
Bottom Line: Googlified Messaging may be a boon to your sex life, but don't stake your job security on it just yet. Also make certain that your voicemail announcement includes a very emphatic request that callers actually identify themselves and leave a callback number where they can be quickly reached.
Google Voice Design. To integrate free voicemail transcription 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.
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.
FreePBX Setup. Obviously there are numerous ways to integrate this transcription service into Asterisk. If you're using FreePBX, here are a couple of simple ways. First, create a Miscellaneous Destination for Google Voice and provide your Google Voice number in the correct format to match your dialplan. Next, if you use a Ring Group to answer incoming calls, choose your new Google Voice Miscellaneous Destination as the "Destination if no Answer." If you're using an IVR to route calls, then perhaps you'll want to add an option to leave a voicemail and have it transcribed for delivery to your email account.
HINT: For rerouting of Asterisk calls to Google Voice, be sure to use an outbound trunk that supports CallerID pass-through. And configure the trunk with a blank CallerID value in FreePBX. Then the actual CallerID of the incoming call will be passed along to Google Voice and stored as part of the voicemail message.
Connecting the Dots. For the visionaries in the audience, you're probably wondering what it would take to add language translation to transcription. So were we. It raises some interesting questions, and some of our early adopters already have tried it. Suffice it to say, it doesn't work yet. But it wouldn't take much effort to run a transcribed message through Google Translate and spit out a Spanish, French, or German message on the other end. Or vice versa: transcribe a German message and translate it into English for email delivery in an English-speaking country. Exciting times, indeed. Stay tuned!
Free U.S. Calls with Google Voice. At least for now, calls through Google Voice to phone numbers in the United States are free. And the rates are quite reasonable to other countries. It's a penny a minute to Canada and two cents a minute to many other countries whose names don't include the word "island." There are several ways to terminate calls through Google Voice with Asterisk. Here's the only way we've found to place outbound calls and also preserve the message transcription functionality.
Log into your Asterisk server as root and 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. Be sure to make the following substitutions to match your Google Voice credentials:
999 - Extension number to call
9876543210 - Your Google Voice 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/9876543210@216.239.37.15:5061
,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 phone number and PIN in the dial string and 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/9876543210@216.239.37.15:5061
,30,mD(wwwwwwwwwwww*ww8888ww2ww${NUM2CALL}#))
exten => *48,n,Hangup
Outbound Trunk Alternative. Since the original article was published, our British colleague, Joe Roper, suggested that we also include instructions for configuring Google Voice as a dial-out trunk (instead of an extension) in Asterisk. The advantage of this approach is that outbound calls can be dialed in the traditional way without interaction with voice prompts. The solution we will outline below lets you place a call 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...
First, log into your Asterisk server as root and edit extensions_custom.conf again. This time, go to the very bottom of the file and add the following code using your Google Voice phone number and PIN. 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.
[custom-google-voice]
exten => _X.,1,Dial(SIP/9876543210@216.239.37.15:5061
,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.
Homework. Google also has introduced a slick new directory assistance service which also is free. We'll leave it to you to take the lesson above and create a GOOG-411 entry in your dialplan. HINT: You choose option 3 instead of option 2 after entering your PIN in the Google Voice menu. Enjoy!
Chapter 2. Google Voice: Is the SIP and Asterisk Honeymoon Over?
Chapter 3. The Return of Googlified Messaging With Free U.S. Calling
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
Free Asterisk Solutions