freepbx – Page 41 – Nerd Vittles

5-Minute VoIP: Deploying a SIP to Google Voice Gateway

Monday, June 11, 2012 / 14 Comments on 5-Minute VoIP: Deploying a SIP to Google Voice Gateway

We’ve been big fans of Google Voice since the outset. But, with the exception of one brief week, the piece Google has always refused to put in place is a SIP gateway to make connections from VoIP devices a no-brainer. You’d think they’d do it for no other reason than economics. SIP calls are free. PSTN calls are not. Well, never mind Google. Bill Simon has done it for you, and he leveraged the same Yate toolkit that Google originally deployed. Today, we’ll show you how to spend five minutes and take advantage of the Simon Telephonics gateway to interconnect a dedicated Google Voice account with any SIP device you’d like, whether it’s an Asterisk® server, a smartphone with a free SIP client from GrooVe IP or Zoiper, a free softphone from Zoiper or X-Lite 4, or any SIP telephone. Once we’re finished today, you can use any SIP client to call your 10-digit Google Voice number through the Simon Telephonics gateway: SIP/9991234567@gvgw1.simonics.com. And you can make and receive calls throughout the U.S. and Canada using your new Google Voice number the old fashioned way, using a Plain Old Telephone. Did we mention that everything is free: the Google Voice number, the Simon Telephonics gateway connection, all of the inbound calls, and outbound calls throughout the U.S. and Canada… at least in 2012. If you take advantage of Bill’s gateway, we would encourage you to at least donate one day’s lunch money to Bill’s site to help pay the light bill.

Getting Started. The drill for today goes like this. First, you’ll create a new Google Voice account with a new phone number at google.com/voice. Next, you’ll make a test call from that number using the Gmail account associated with that same account. Then, you’ll register the Google Voice number on the Simon Telephonics gateway. Next, we’ll set up a SIP trunk on your Asterisk server for this new DID. Finally, configure any SIP client with an extension number from your Asterisk PBX, and you can start making and receiving calls using your new Google Voice number.

A Word About Security. Google doesn’t (yet) support OAuth authentication for Google Voice accounts. What this means is that you’ll have to use your actual Google Voice credentials to set up your account on the Simon Telephonics gateway. Could Bill steal your credentials? Absolutely. Will he? Absolutely not. Why? First, there’s no money in your Google Voice account so all he could do is make free calls on Google’s nickel, the same thing he could do using his own Google Voice accounts. Second, Bill is better off setting up his own accounts where you don’t share his password and the Google Voice call logs won’t tell you who he’s calling. If you’re paranoid, don’t put money in your calling account, make the account name something that could not be associated with you, and then check your call logs several times every day. Better yet, spend $50 and use an OBi110 device to set up your own private gateway where Obihai knows your credentials instead of Bill. 😉

Configuring Google Voice. As we mentioned, you’ll need a dedicated Google Voice account for this. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now.

We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively for this new SIP gateway. Head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for the SIP gateway to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

Call Screening – OFF
Call Presentation – OFF
Caller ID (In) – Display Caller’s Number
Caller ID (Out) – Don’t Change Anything
Do Not Disturb – OFF
Call Options (Enable Recording) – OFF
Global Spam Filtering – ON

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

Finally, go into Gmail for this same account and place a test call using your new Google Voice number. You’ll find the Call Phone icon in the Chat and SMS section of Gmail in the left column. Once you complete this step, be sure to log out of both Gmail and Google Voice for this account, or inbound calling will never work.

Registering on the Simon Telephonics Gateway. Now we’re ready to register your Google Voice account on the Simon Telephonics Gateway. Click on the link and fill in the blanks with your Google Voice account credentials and phone number. Be sure to include a 1 at the beginning of your Google Voice number! You’ll note that Google Apps email domains are supported as well as gmail.com addresses.

Google Voice Number – 19991234567
GV Username – joeschmo2468
GV Domain – gmail.com
GV Password – mightysecret
GV Password again – mightysecret
Email Address – joeschmo@yahoo.com

Check your entries carefully and then click the Add button. The only way to make changes if you screw things up is to delete the existing account by entering your original credentials to Delete the original account and then you Add a new one. So type carefully and check your work. Once your account is successfully registered, the Simon Telephonics Gateway will spit back your new SIP credentials. Write them down or take a screenshot and put them in a safe place. You’ll need them to set up your Asterisk SIP trunk. The Username will be your 11-digit Google Voice number with a GV prefix. The Secret will be a randomized string. The Registration String will be used in setting up your Asterisk SIP trunk and is in the proper format. The DID for your Inbound Route in FreePBX® will be your 11-digit Google Voice number.

Server – gvgw1.simonics.com
Username – GV19991234567
Secret – Xyzkk
Registration String – GV19991234567:Xyzkk@gvgw1.simonics.com/19991234567
Dialing Format – E.164 without + (for US calls, 11 digits starting with 1)

NOTE: Newer users may be provided an alternate gateway, e.g. gvgw2.simonics.com. You would obviously need to use whichever gateway FQDN is provided in all of the settings shown here.

Creating FreePBX SIP Trunk. Now we’re ready to create your new SIP trunk in FreePBX. Choose Add SIP Trunk and fill in the blanks as shown below with your new credentials. The Trunk Name can be any name you like. Don’t forget the 1 in Prepend for the Dialed Number Manipulation Rules! Leave the Incoming Settings blank. Be sure to add your Registration String from the credentials that were provided as part of the Simon Telephonics registration. Then Save Your Settings.

Creating FreePBX Inbound Route. Now you’ll need to add an Inbound Route to process incoming calls from the Simon Telephonics Gateway. The DID entry will be your 11-digit Google Voice number. The Destination for the incoming calls can be whatever you like: an extension, a ring group, an IVR, or any of the other available options on your server.

Creating FreePBX Outbound Route. If you want to send outbound calls out through your new Google Voice trunk, then you’ll need to add the SIP trunk to your outbound dialing rules. Just add the SIP Trunk Name you’ve defined to the Trunk Sequence for calls with the NXXNXXXXXX Dial Pattern, and you’re all set. Enjoy!

Originally published: Monday, June 11, 2012

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

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…

Eating Our Own Dog Food: And It Tastes Pretty Good

Monday, June 4, 2012

Original Dinner Time photo courtesy of April Turner

Frequent visitors to Nerd Vittles know that we’re pretty vocal about the cardinal rule that software developers should eat their own dog food before foisting it on the unsuspecting public. Another company that shares our view is Schmooze Com®, the primary developers of the latest releases of FreePBX®.

They say a picture is worth a thousand words, and we wanted to share a photo from the Schmooze "phone lab" where every release of FreePBX is put through its paces. It also serves as a primary support tool for the Schmooze Tech Support team when folks finally decide they need immediate help with a technical problem, and time is of the essence.

PBX in a Flash™ continues to offer a rock-solid reliable VoIP telephony platform primarily because its developers use it and refine it through real-world deployments. As some of you may have heard, we’ve actually decided to write a real book about PBX in a Flash, and the very first thing we decided to do was to build and deploy a new server using the latest and greatest of everything so that every reader could rest assured that anything suggested in our book was not only being tested but actually being used in a production environment. That includes the latest in virtual private networks, text-to-speech and speech-to-text applications, Google Voice, iNum, connections to a dozen of our favorite VoIP providers, and healthy workouts for our favorite phones including the Nortel 1535 IP color videophone and the Aastra and Digium business phone sets. We also plan to write about our user community and the PIAF Forums, both of which are really second to none. They’re an integral part of why PBX in a Flash works as well as it does because we have literally hundreds of gurus that also eat the dog food to make sure the developers haven’t missed something.

As ancient technology veterans, we’ve continued to search for the perfect fit for PIAF’s missing piece, a commercial support option for organizations that really, truly need 24×7 support when something comes unglued. If your business depends upon telephones (and most do), then Tier 3 support isn’t a luxury. It’s a necessity. If your in-house technology team is spread thin or enjoys an occasional vacation without worrying whether Rome will burn while you’re out cruising the Mediterranean, then commercial support as a backup is absolutely essential. And, heaven forbid, there might be something you actually don’t know about your phone system that could take it to an entirely different level. PBX in a Flash now has been deployed by major airlines, by huge government organizations, and throughout the commercial sector to support every type of business you can imagine. We also have a growing list of system integrators that deploy PBX in a Flash for their commercial clients. All of these folks need a reliable, knowledgable, available technology expert not only as a backup but also to address the unexpected. Now we’ve got it!

Philippe Lindheimer of FreePBX and Tony Lewis of Schmooze Com have been terrific business partners of ours since before there was a PBX in a Flash. Many years ago, we personally encouraged Philippe to release an aggregation so that we wouldn’t have to. Tony was equally reluctant because of his commercial deployment responsibilities. Suffice it to say, we all saw the writing on the wall with the morphing of Asterisk@Home into trixbox®. Times have changed! Fast forward five years and we have a new VoIP World. Tony’s business has grown astronomically and his commercial support of those using FreePBX as the premier web management GUI for Asterisk® is legendary.

We are thrilled to announce that Schmooze Com now is the official, commercial support organization for all PBX in a Flash systems. They will work closely with you and with the PIAF Dev Team to promptly address and resolve any problem that demands immediate attention. While the PIAF Forums will continue to offer first-class, free support for all of our users, we now have a commercial solution as well. Whether you’re a one-person Mr. Fixit or part of a large IT support organization, we think you’ll find today’s commercial support offering a welcome addition.

It’s incredibly easy to sign up and take advantage of the new service. Just visit pbxinaflash.com/support and fill out the form to set up a commercial account. Once completed, you can take advantage of the new service whenever you need it. We’re confident that it’ll take your PBX in a Flash system to a whole new level and make a lot of you that do this for a living sleep better. As always, we’re as close as your email inbox if you have comments or suggestions about the new service. Enjoy!

Free Conference Bridge. We’re delighted to announce that les.net now is offering a free, 100-user conference bridge to all of its customers. You can read all about it on the PIAF Forums.

Originally published: Monday, June 4, 2012

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

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…

PIAF 2.0.6.2.3: It’s PIAF-Brown with Certified Asterisk

Monday, May 7, 2012 / 5 Comments on PIAF 2.0.6.2.3: It’s PIAF-Brown with Certified Asterisk

With apologies to UPS: Any likeness of this truck to the brown trucks that stop by our office every afternoon is purely coincidental.

Today we're delighted to take Asterisk® to the next plateau with an all-new release of PBX in a Flash™. Tom King's latest masterpiece gives you unparalleled flexibility with the ease of installation and security you've come to expect from all PIAF™ releases. Featuring CentOS® 6.2, this new release provides your choice of the most stable versions of Asterisk 1.8 or 10 and now includes the option to install the new Certified Asterisk release from Digium® featuring SLA support for the first time. Certified Asterisk also brings plug-and-play support for Digium Phones. And, of course, you still get your choice of FreePBX® 2.8, 2.9, or 2.10 as well as one-click installs for Incredible PBX and Incredible Fax. No other platform gives YOU this kind of flexibility to easily design a telephony platform that meets all your unique requirements.

We're also pleased to announce that, in addition to the one-of-a-kind PBX in a Flash Forum, we're adding what many of you have requested, a paid support service from the best folks in the business for organizations that truly need immediate technical assistance when something comes unglued. More details are coming in the next few weeks so stay tuned.

Featuring superior scalability, improved performance, better resource management, and unmatched device support, PBX in a Flash 2.0.6.2 brings you the most versatile Asterisk platform on the planet with the latest and greatest releases of virtually every major open source product in the marketplace. And you can choose either the 32-bit or 64-bit platform. For those needing additional Asterisk customization, PIAF2 also provides direct access to Asterisk's menuconfig system which lets you tailor the selection of Asterisk modules you wish to deploy. With 2.0.6.2.3, we've included a new menu-driven installation option that lets you add network drivers for many of the latest and greatest network cards that are not yet supported in the Linux kernel. We've also updated the kernel to eliminate many of the bugs reported in the default CentOS 6.2 kernel. And, of course, PIAF2 continues to provide the only turnkey Google Voice solution providing immediate free calling throughout the U.S. and Canada with any of the default flavors of PIAF.

Here's how the new install works. Step #1 is downloading the ISO and burning a CD (32-bit) or DVD (64-bit) to install your new server. If your machine lacks a CD/DVD drive, there's now a simple procedure for building a USB Flash Drive installer. Once you boot your server using the installer you created, you first get to select the file system for your new CentOS server. For most folks, just press the Enter key. Next, you'll be prompted to create a very secure root password. Then the PIAF2 disk will whir away for about 15 minutes installing CentOS 6.2. As your system reboots, remove the install disk and the second phase of the install begins.

The PIAF2 Install Menu will display. It looks like this:

Adding a Network Driver. In addition to choices to select your flavor of Asterisk to install, the main install menu now includes an option to load Add-On network drivers. With Linux, network drivers have to be part of the Linux kernel to work. And you must have Internet connectivity to complete the Phase II install of PBX in a Flash. So this eliminates the chicken-and-the-egg problem. As new hardware appears, the latest drivers haven't yet made it into the kernel. So we now let you add the one you need from the Extra Drivers Menu. It's literally plug-and-play. Click the driver you need, and presto it's added. Reboot to return to the Install Menu and continue your install. By the way, if you don't know what type network card you have in your system, drop down to the Linux CLI, log in as root, and type: lspci. Then continue by entering the command: piafdl.

Unlike other distros, we build your server the old-fashioned way, from source, and then compile the various components. This gives you the flexibility to add new features and recompile your applications down the road when new additions become available that you'd like to include in your system. It also provides the flexibility to adjust your Asterisk setup to meet your specific requirements. In short, it's just as if you'd installed Asterisk manually from source code. Why? Because you have!

Choosing a Custom Version of Asterisk. From the Main Install Menu, you also have the option of exiting to the Linux command prompt to select from a broad list of newer Asterisk releases. If you choose this option, you'll be prompted to log into your server as root with the root password you chose initially. Once logged in, you can execute any series of Linux commands or issue one of the following commands to choose a specific release of Asterisk:

piafdl -p beta_1880_purple (loads Asterisk 1.8.8.0)
piafdl -p beta_1881_purple (loads Asterisk 1.8.8.1)
piafdl -p beta_1882_purple (loads Asterisk 1.8.8.2)
piafdl -p beta_1890_purple (loads Asterisk 1.8.9.0)
piafdl -p beta_1891_purple (loads Asterisk 1.8.9.1)
piafdl -p beta_1892_purple (loads Asterisk 1.8.9.2)
piafdl -p beta_1893_purple (loads Asterisk 1.8.9.3)
piafdl -p beta_18101_purple (loads Asterisk 1.8.10.1)
piafdl -p beta_18110_purple (loads Asterisk 1.8.11.0)
piafdl -p beta_18111_purple (loads Asterisk 1.8.11.1)
piafdl -p beta_18120_purple (loads Asterisk 1.8.12.0)
piafdl -p beta_1811_brown (loads Asterisk 1.8.11-cert1)
piafdl -p beta_1000_red (loads Asterisk 10.0.0)
piafdl -p beta_1001_red (loads Asterisk 10.0.1)
piafdl -p beta_1010_red (loads Asterisk 10.1.0)
piafdl -p beta_1011_red (loads Asterisk 10.1.1)
piafdl -p beta_1012_red (loads Asterisk 10.1.2)
piafdl -p beta_1013_red (loads Asterisk 10.1.3)
piafdl -p beta_1020_red (loads Asterisk 10.2.0)
piafdl -p beta_1021_red (loads Asterisk 10.2.1)
piafdl -p beta_1030_red (loads Asterisk 10.3.0)
piafdl -p beta_10311_red (loads Asterisk 10.3.1.1)
piafdl -p beta_1040_red (loads Asterisk 10.4.0)

Be advised that Asterisk 10.1.x and later releases reportedly break Google Voice! The good news is that the new PIAF deployment methodology for newer Asterisk releases is working. We no longer incorporate the latest Asterisk release as the default PIAF install. Instead, you get a version that has undergone thorough independent testing by our gurus. So the base PIAF install of Asterisk 10 still gets you a version of Asterisk that reliably supports Google Voice.

Picking a Flavor of Asterisk. Many of you already know which branch of Asterisk you prefer to install, but some don't. If there's not a particular reason for choosing PIAF-Red with Asterisk 10, don't. The reason is that support for it ends within the year. PIAF-Purple with Asterisk 1.8 on the other hand will be supported with bug fixes for several more years. And, if you value an SLA (meaning reported bugs will get fixed) or you want to use Digium Phones, then PIAF-Brown is your only option at the moment. Even though it is new, it is based upon a stable version of Asterisk 1.8. We're running it with good results. YMMV!

The Config Module. Once you choose your flavor of Asterisk, the PIAF2 install will continue by loading the Config Module. Within this module, you do the following:

First, you get to choose whether to access the Asterisk menuconfig utility which lets you select which modules of Asterisk to install. Don't pick this option unless you know what you're doing and need something special. We've enabled all of the options that most folks need.

Second, you'll be prompted to choose your Time Zone. Choose from one of the options provided, and press Y to confirm your choice.

Third, you'll be prompted to choose your flavor of FreePBX to install: 2.8, 2.9, or 2.10. This choice is important. If you want to use Incredible PBX or Incredible Fax or both, then you must select either FreePBX 2.9 or 2.10. If you want to use the FreePBX Digium Phones module, then your only choice is FreePBX 2.10. While it's possible to upgrade FreePBX to a later version in place, it's not painless so make your selection carefully.

Fourth, you must assign a password for access to the FreePBX GUI and utilities. Make it secure or let the system pick one for you. It's your phone bill.

Finally, press Enter to confirm your selections. Then go have a cup of coffee. The install process will continue for 15-30 minutes depending upon the speed of your server and network connection. The necessary components that you've chosen will be downloaded and compiled. You'll also get an updated CentOS 6.2 system as all of the yum updates are applied for your server.

When the install finishes, your system will reboot a final time and then you'll have a working PIAF2 server.

What Next? Before you make any changes using FreePBX, you have a few decisions to make. If you plan to use Incredible PBX and/or Incredible Fax, you need to install them now in the order shown because they overwrite all of your FreePBX settings. Now would be a good time to read the Nerd Vittles article which explains the functionality and installation process for these two great products.

If you want to use Incredible PBX, install it first! Log in to the Linux CLI and issue the command: install-incredpbx3.

If you want free faxing support for your PIAF2 server, install it after Incredible PBX by issuing the command: install-incredfax2.

If you are using PIAF-Brown with FreePBX 2.10 and want to take advantage of the FreePBX Digium Phones module (shown above), now's the time to install it after the other two installs above:

Step #1: Obtain a free Digium DPMA license key for your server.

Step #2: Log in as root and issue the command: install-digiphones.

FreePBX Setup. Most of your time with PIAF2 will be spent using a browser and the FreePBX GUI. To get to it, you'll need the IP address of your server. Log into the Linux CLI as root using your root password. Write down the IP address of your server from the status display (below) and verify that everything installed properly. Note that Samba is disabled by default. If you want to use your server with Windows Networking, run configure-samba once your server is up and running.

HINT: Run status from the Linux CLI to decipher the IP address of your server.

Once you have the IP address of your server, just point your browser to that IP address to bring up the PIAF GUI (shown below). Review the PIAF RSS Feed. We recommend checking this RSS Feed daily by pointing your browser to the IP address of your server. The RSS Feed is displayed in the left column of the GUI and will alert you to any newly discovered security vulnerabilities in CentOS, Asterisk, FreePBX, or PIAF2. Click on the Users tab to change to the Admin panel, and then select FreePBX to load the FreePBX GUI.

You also can access the FreePBX GUI directly by pointing your browser to the IP address of your PIAF2 server: http://ipaddress/admin. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in the Config Module phase of the PIAF2 install above.

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

A Word About Security. PBX in a Flash has been engineered to run on a server sitting safely behind a hardware-based firewall with NO port exposure from the Internet. Leave it that way! It's your wallet and phone bill that are at stake. If you're running PBX in a Flash in a hosted environment with no hardware-based firewall, then immediately read and heed our setup instructions for Securing Your VoIP in the Cloud Server.

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

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

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

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

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

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

Courtesy of wordle.net

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

For today, we're going to take advantage of Google's current offer of free calling in the U.S. and Canada through the end of 2012. You also get a free phone number in your choice of area codes. PBX in a Flash now installs a Google Voice module for FreePBX that lets you set up your Google Voice account with PBX in a Flash in just a few seconds once you have your credentials.

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

We've tested this extensively using an existing Gmail account rather than creating a separate account. Take our word for it. Inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So... set up a dedicated Gmail and Google Voice account¹, and use it exclusively with PBX in a Flash. Google Voice no longer is by invitation only. If you're in the U.S. or have a friend that is, head over to the Google Voice site and register. If you're living on another continent, see MisterQ's posting for some tips on getting set up.

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

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

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

Call Screening - OFF
Call Presentation - OFF
Caller ID (In) - Display Caller's Number
Caller ID (Out) - Don't Change Anything
Do Not Disturb - OFF
Call Options (Enable Recording) - OFF
Global Spam Filtering - ON

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

Configuring Google Voice Trunk in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize PBX in a Flash to meet your specific needs. Click the Setup tab and choose Google Voice in the Third Party Addons. To Add a new Google Voice account, just fill out the form:

Phone number is your 10-digit Google Voice number. Username is your Google Voice account name without @gmail.com. NOTE: You must use a Gmail.com address in the current version of this module! Password is your Google Voice password. NOTE: Don't use 2-stage password protection in this Google Voice account! Be sure to check all three boxes: Add trunk, Add routes, and Agree to TOS. Then click Submit Changes and reload FreePBX. Down the road, you can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill. For Google Apps support, see this post on the PIAF Forum.

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

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

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

General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. Scroll to the bottom of the General Settings screen after selecting it from the left panel. Plug in your email address, click Submit, and save your changes. Done!

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you'll want a real SIP telephone such as the $50 Nortel color videophone we've recommended previously. You'll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you're like us, we want to make damn sure this stuff works before you shell out any money. So, for today, let's download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 201 for your extension and your actual password for extension 201. Then plug in the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Enabling Google Voicemail. Some have requested a way to retain Google's voicemail system for unanswered calls in lieu of using Asterisk voicemail. The advantage is that Google offers a free transcription service for voicemail messages. To activate this, you'll need to edit the [googlein] context in extensions_custom.conf in /etc/asterisk. Just modify the last four lines in the context so that they look like this and then restart Asterisk: amportal restart

;exten => s,n(regcall),Answer
;exten => s,n,SendDTMF(1)
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => s,n,Goto(from-trunk,gv-incoming,1)

But I Don't Want to Use Google Voice. If you'd prefer not to use Google Voice at all with PBX in a Flash, that's okay, too. Here's how to disable it and avoid the chatter in the Asterisk CLI. Log into your server as root and edit /etc/asterisk/modules.conf. Change the first three lines in the [modules] context so that they look like this. Then restart Asterisk: amportal restart.

autoload=yes
noload => res_jabber.so
noload => chan_gtalk.so

There's now a patch that automatically adjusts Asterisk to accommodate Google Voice whenever you have added Google Voice extensions to your system. To download and install the patch, visit the PIAF Forum.

Originally published: Monday, May 7, 2012

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.

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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest...

You also can use a dedicated Googtle Apps account for Google Voice with the latest version of the FreePBX module. Don't use your regular Google Apps email address with Google Voice, or inbound calling will not work! [↩]

Introducing PBX in a Flash 2 with CentOS 6.2

Friday, January 20, 2012 / 2 Comments on Introducing PBX in a Flash 2 with CentOS 6.2

Today we're delighted to introduce the ultimate Asterisk® platform. It's the all new PBX in a Flash 2™ featuring CentOS® 6.2 and your choice of Asterisk 1.8.8.0 or 10 plus FreePBX® 2.8, 2.9, or 2.10. No other platform gives YOU the flexibility to design a telephony platform that meets your unique requirements. And, of course, no other platform includes any version of CentOS 6, much less 6.2.

Featuring superior scalability, improved performance, better resource management, and unmatched device support, PBX in a Flash 2.0.6.2 brings you the most versatile Asterisk platform on the planet with the latest and greatest releases of virtually every major open source product in the marketplace. And you can choose either the 32-bit or 64-bit platform. For those needing additional Asterisk customization, PIAF2 also provides direct access to Asterisk's menuconfig system which lets you tailor the selection of Asterisk modules you wish to deploy. And, of course, PIAF2 continues to provide the only turnkey Google Voice solution providing immediate free calling throughout the U.S. and Canada. We'll walk you through the 2-minute drill to deploy Google Voice for inbound and outbound calling with FreePBX. And, yes, Incredible PBX 2.9 is fully compatible with the 32-bit release of PIAF 2.0.6.2!

Our special tip of the hat again goes to Tom King, who has spent the better part of four months integrating PIAF2 into the new CentOS 6 releases, three of them to be exact. To suggest that this was not a job for mere mortals doesn't begin to paint the picture of this long and winding road. The good news is we think you'll be delighted with the results. The PBX in a Flash install process now has been streamlined into three distinct components.

After downloading the ISO and burning a CD (32-bit) or DVD (64-bit) to install your new server, here's how it works. First, you get to choose the file system for your new CentOS server. The PIAF2 installer will whir away for about 15 minutes installing CentOS 6.2. When your system reboots, remove the install disk and Phase 2 begins. Here you get to choose your flavor of Asterisk to deploy. We continue to recommend PIAF2-Purple as the stable product for all but pioneers, but Asterisk 10 is out of beta, and we offer you the option of installing it if you wish.

By default with PIAF2, you get your choice of Asterisk 1.8.8.0 or 10.0.0 as well as FreePBX 2.8, 2.9, or 2.10. With the standard PIAF2 ISO installer, you also have the option of exiting to the Linux command prompt to compile a network driver or to select from a broad selection of newer Asterisk releases. If you choose this option, you'll be prompted to log into your server as root with the root password you chose initially. Once logged in, you can execute any series of Linux commands or issue one of the following commands to choose a specific release of Asterisk:

piafdl -p beta_1880_purple (loads Asterisk 1.8.8.10)
piafdl -p beta_1881_purple (loads Asterisk 1.8.8.1)
piafdl -p beta_1882_purple (loads Asterisk 1.8.8.2)
piafdl -p beta_1890_purple (loads Asterisk 1.8.9.0)
piafdl -p beta_1891_purple (loads Asterisk 1.8.9.1)
piafdl -p beta_1892_purple (loads Asterisk 1.8.9.2)
piafdl -p beta_1893_purple (loads Asterisk 1.8.9.3)
piafdl -p beta_1001_red (loads Asterisk 10.0.1)
piafdl -p beta_1010_red (loads Asterisk 10.1.0)
piafdl -p beta_1011_red (loads Asterisk 10.1.1)
piafdl -p beta_1012_red (loads Asterisk 10.1.2)
piafdl -p beta_1013_red (loads Asterisk 10.1.3)

WARNING: Asterisk 10.1.x releases reportedly break Google Voice! The good news is that the new PIAF deployment policy for Asterisk releases is working. We no longer incorporate the latest Asterisk releases as the default PIAF install before independent testing. You, of course, are free to load and test any of the releases you wish using the commands outlined above.

If you compiled a network driver and wish to resume the installation process, just reboot the server. If you chose a specific flavor of Asterisk, simply accept the license agreement and the customized PIAF2 install will continue.

Within a minute or so, your chosen Asterisk installer will load. In Phase 3 (the Config Module), you'll pick your flavor of FreePBX and choose a password for access, set your time zone, and decide whether you want to further customize Asterisk using menuconfig.

If you want to also install Incredible PBX 2.9, be sure to use the 32-bit PIAF2 ISO and choose Asterisk 1.8 and FreePBX 2.9.

Otherwise, the choices are up to you. Once you've made your selections, everything else installs on autopilot unless you opted to use menuconfig. If so, come back in 15 minutes and tailor away. Then press x to save your settings and finish the install. Depending on the speed of your server or virtual machine, the complete install usually takes 30-60 minutes. It's not the fastest Asterisk install on the planet. But, as you learned in high school, faster isn't always better. With PIAF2, you get a fully customized Asterisk environment with the very latest CentOS 6.2 updates.

After the final reboot, you'll have a working PIAF2 server. Open up FreePBX with a browser, enter your Google Voice credentials, create an extension, link an inbound route to that extension to accept calls, restart Asterisk from the command prompt, and you'll have a fully operational PBX in less than 2 minutes.

Creating a PIAF2 Install Disk. To get started, download the PIAF2 ISO of your choice from SourceForge.

Once you have the ISO image in hand, the next step is to burn the ISO image to a DVD. The 32-bit ISO still will fit on a CD if you prefer. If you've never done it before, here's a DVD tutorial that will show you how on either a Windows machine or a Mac. If your machine lacks a CD/DVD drive, there's now a simple procedure for building a USB Flash Drive installer.

Using PIAF2 with Proxmox. For those using Proxmox to host PIAF2 virtual machines, the easiest approach is to log into your server as root, change to the /var/lib/vz/template/iso directory, and issue a wget command to download the SourceForge image of your choice. In building KVM virtual machines with Proxmox, you'll need to allocate at least 768MB of RAM (1024MB recommended) for each image. CentOS 6 has a much larger memory footprint than CentOS 5. Reminder: Be absolutely sure Proxmox is sitting behind a secure hardware-based firewall. It is NOT secure on the open Internet!

Atom-based PC Platform. Unless you're using PIAF2 on a virtual hosting platform, you'll need a dedicated PC. For the least expensive hardware alternative, pick up an Atom-based PC. We previously have recommended against an EEE PC because of the network driver incompatibility with CentOS 5. We'll have to leave it to the pioneers to tell us whether this still applies with CentOS 6. We do know that the refurbished Acer desktops work fine. Someone has actually tested them! And they can easily support a small business with dozens of phones. See these performance benchmarks for details.

Another terrific option (if you hurry) is this refurbished Dell GX620 for $79.99. These won't last long.
PIAF Web GUI with RSS Feed

FreePBX Setup. After the PIAF2 install finishes, your server will reboot once again. Log into the Linux CLI as root using your root password. Write down the IP address of your server from the status display and verify that everything installed properly. Note that Samba is disabled by default. If you want to use it for Windows Networking, run configure-samba once your server is up and running.

Most of your life with PBX in a Flash will be spent using the FreePBX web GUI and your favorite browser. Just point your browser to the IP address of your server and review the PIAF RSS Feed (as shown above). We recommend checking this RSS Feed daily by pointing your browser to the IP address of your server. The RSS Feed is displayed in the left column of the GUI and will alert you to any newly discovered security vulnerabilities in CentOS, Asterisk, FreePBX, or PIAF2. Click on the Users tab to change to the Admin panel, and then select FreePBX to load the FreePBX GUI.

You also can access the FreePBX GUI directly by pointing your browser to the IP address of your PIAF2 server: http://ipaddress/admin. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in Phase 3 of the PIAF install.

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

A Word About Security. PBX in a Flash has been engineered to run on a server sitting safely behind a hardware-based firewall with NO port exposure from the Internet. Leave it that way! It's your wallet and phone bill that are at stake.

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

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

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

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

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

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

Courtesy of wordle.net

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

For today, we're going to take advantage of Google's current offer of free calling in the U.S. and Canada through the end of 2012. You also get a free phone number in your choice of area codes. PBX in a Flash now installs a Google Voice module for FreePBX that lets you set up your Google Voice account with PBX in a Flash in just a few seconds once you have your credentials.

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

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

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

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

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

Call Screening - OFF
Call Presentation - OFF
Caller ID (In) - Display Caller's Number
Caller ID (Out) - Don't Change Anything
Do Not Disturb - OFF
Call Options (Enable Recording) - OFF
Global Spam Filtering - ON