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The Most Versatile VoIP Provider: FREE PORTING

VPN in a Flash Reborn: Meet the Dedicated Server Edition in PIAF 2.0.6.2.4

We’re dusting off our favorite old trademark to introduce the all-new VPN in a Flash™ featuring NeoRouter™ 1.7 Free Server Edition. Last month we showed how to install NeoRouter as an add-on for existing PBX in a Flash™ servers. In sites with 10 or fewer machines to interconnect, this works extremely well. However, for those with major collections of servers and PCs scattered across the universe (up to 256!), you’re going to want dedicated hardware to manage your virtual private network. Thanks to the terrific work of Tom King, you’ve got that choice. Meet VPN in a Flash.

As with PBX in a Flash, the Dedicated Server Edition of VPN in a Flash is offered in 32-bit or 64-bit flavors. How do you get it? It’s now an option in the PBX in a Flash 2.0.6.2.4 ISO featuring the CentOS 6.2 platform for the ultimate in reliability. Just download the new 2.0.6.2.4 ISO from SourceForge, burn it to a CD or DVD or, better yet, make yourself a bootable flash drive, and find some hardware to dedicate to the task of managing your virtual private network. Set up the server behind a dedicated firewall on any private LAN other than the 10.0.0.x network. Answer a few prompts to choose your timezone and set up your NeoRouter credentials. Then configure your hardware firewall to lock down the assigned DHCP address of your VPN in a Flash server and map TCP 32976 to the IP address of your VPN server, and you’re done. In 30 minutes, you get a rock-solid, preconfigured VPN. Not only is it SECURE, it’s also FREE!

After your VPN in a Flash server is installed, you can optionally go to the NeoRouter web site and register your new VPN by clicking Create Standalone Domain. Make up a name you can easily remember with no periods or spaces. You’ll be prompted for the IP address of your server in the second screen. FQDNs are NOT permitted.

When a VPN client attempts to login to your server, the server address is always checked against this NeoRouter database first before any attempt is made to resolve an IP address or FQDN using DNS. If no matching entry is found, it will register directly to your server using a DNS lookup of the FQDN. Whether to register your VPN is totally up to you. Logins obviously occur quicker using this registered VPN name, but logins won’t happen at all if your server’s dynamic IP address changes and you’ve hard-coded a different IP address into your registration at neorouter.com.

Setting Up a NeoRouter Client. There are NeoRouter clients available for almost every platform imaginable, except iPhones and iPads. Hopefully, they’re in the works. So Step #1 is to download whatever clients are appropriate to meet your requirements. The VPN in a Flash install automatically loads the Linux clients into the /usr/src/neorouter directory and installs the NeoRouter client for you. Here’s the NeoRouter Download Link for the other clients. Make sure you choose a client for the Free version of NeoRouter. And make sure it is a version 1.7 client! Obviously, the computing platform needs to match your client device. The clients can be installed in the traditional way with Windows machines, Macs, etc.

CentOS NeoRouter Client. As part of the installation above, we have automatically installed the NeoRouter client for your particular flavor of CentOS 6, 32-bit or 64-bit. In order to access resources on your NeoRouter server from other clients, you will need to activate the client on your server as well. This gets the server a private IP address in the 10.0.0.0 network.

To activate the client, type: nrclientcmd. You’ll be prompted for your Domain, Username, and Password. You can use the registered domain name from neorouter.com if you completed the optional registration step above. Or you can use the private IP address of your server. If your router supports hairpin NAT, you can use the public IP address or server’s FQDN, if you have one. After you complete the entries, you’ll get a display that looks something like this:

To exit from NeoRouter Explorer, type: quit. The NeoRouter client will continue to run so you can use the displayed private IP addresses to connect to any other online devices in your NeoRouter VPN. All traffic from connections to devices in the 10.0.0.0 network will flow through NeoRouter’s encrypted VPN tunnel. This includes inter-office SIP and IAX communications between Asterisk® endpoints.

Admin Tools for NeoRouter. Here are a few helpful commands for monitoring and managing your NeoRouter VPN.

Browser access to NeoRouter Configuration Explorer (requires user with Admin privileges)

Browser access to NeoRouter Network Explorer (user with Admin or User privileges)

To access your NeoRouter Linux client: nrclientcmd

To restart NeoRouter Linux client: /etc/rc.d/init.d/nrservice.sh restart

To restart NeoRouter Linux server: /etc/rc.d/init.d/nrserver.sh restart

To set domain: nrserver -setdomain YOUR-VPN-NAME domainpassword

For a list of client devices: nrserver -showcomputers

For a list of existing user accounts: nrserver -showusers

For the settings of your NeoRouter VPN: nrserver -showsettings

To add a user account: nrserver -adduser username password user

To add admin account: nrserver -adduser username password admin

Test VPN access: http://www.neorouter.com/checkport.php

For a complete list of commands: nrserver –help

To change client name from default pbx.local1:

  • Edit /etc/hosts
  • Edit /etc/sysconfig/network
  • Edit /etc/sysconfig/network-scripts/ifcfg-eth0
  • reboot

For the latest NeoRouter happenings, follow the NeoRouter blog on WordPress.com.

Eating Our Own Bear Food. We’ve actually been at our SOHO cabin this month "testing" VPN in a Flash. It’s provided instant access both to our desktop machines and servers in Charleston as well as Tom King’s Proxmox server in Florida where we’ve been developing Yate in a Flash™, a new, dedicated SIP to Google Voice Gateway for Asterisk. We’ll have more to say about it next week, or you can follow the link and get a head start. The bottom line on VPN in a Flash: It Just Works! VPN in a Flash frees you from ever having to stay in your home or office to get work done. And it’s been rock-solid reliable. Enjoy!

Originally published: Wednesday, June 20, 2012



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


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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…

  1. We’ve built a script to rename your VPN in a Flash server in all the right places. You can download it here. []

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

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.

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.

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...

  1. 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 NeoRouter VPN: A Star Is Born

In our last article, we introduced PPTP VPNs for interconnecting remote users and branch offices to a central network hub. Known as a hub-and-spoke VPN, the advantage of this design is it lets remote users participate as peers in an existing home office LAN. It’s simple to set up and easy to maintain. The drawback is vulnerability to man-in-the-middle attacks.

Today, we want to turn our attention to the more traditional client-server VPN which still relies upon a central server but uses a star topology to connect remote nodes. The major difference is that only registered devices participate in the virtual private network so there is no direct access to other machines on the LANs of the registered devices. If you have servers scattered all over the countryside, this is an excellent way to manage and interconnect them. All data and communications between the nodes can then be routed through the encrypted VPN tunnel for rock-solid security.

With NeoRouter’s free software, you can set up your VPN server using a PC, a Mac, a Linux or FreeBSD machine, OpenWrt Backfire, and Tomato. VPN clients are available for PCs, Macs, Linux and FreeBSD PCs, OpenWrt, Tomato as well as Android phones and tablets. There’s even an HTML5 web application in addition to a Chrome browser plug-in. With the OpenWrt and Tomato devices or if you’re an extreme techie, you can broaden your NeoRouter star configuration to include bridging of remote LANs. See pp. 47-50 of the NeoRouter User’s Manual. And you can interconnect up to 256 devices at no cost. For $999, you can enlarge your VPN to support 1,000 devices. Screen sharing, remote desktop connections, HTTP, and SSH access all work transparently using private IP addresses of the VPN nodes which are automatically assigned to the 10.0.0.0 private network.

You may be wondering why we’ve moved on from Hamachi. Suffice it to say, LogMeIn has put the squeeze on the free version to the point that it’s now next to worthless. In fact, you’d be hard-pressed to find any mention of a free version of Hamachi (other than a trial edition) on LogMeIn’s current web site. Here’s a feature comparison which says it better than we could.

Today we are introducing the first of two NeoRouter VPN solutions. First, we have a simple installation script that works with any PBX in a Flash 2™ server. See also our more recent column for the dedicated server edition of NeoRouter VPN known as VPN in a Flash. It’s suitable for use on a dedicated server or running as a virtual machine. For smaller VPNs, we prefer the add-on module for PBX in a Flash. For larger deployments, you probably should opt for the dedicated machine. It also isolates your VPN server from your PBX which generally is the better network strategy. Regardless of the installation scenario you choose, keep in mind that neither option requires exposure of your entire server to the Internet. Only a single TCP port needs to be opened in your hardware-based firewall and IPtables Linux firewall.

NeoRouter Setup with PIAF2™. We’re assuming you already have a PBX in a Flash 2 server set up behind a hardware-based firewall. If not, start there. Next, we’ll need to download and run the installer for your new NeoRouter Server. It also installs the client. Just log into your server as root and issue the following commands:

wget http://incrediblepbx.com/install-neorouter
chmod +x install-neorouter
./install-neorouter

The installer will walk you through these five installation steps, but we’ll repeat them here so you have a ready reference down the road.

First, on your hardware-based firewall, map TCP port 32976 to the private IP address of your PIAF2 server. This tells the router to send all NeoRouter VPN traffic to your PIAF2 server when it hits your firewall. If you forget this step, your NeoRouter VPN will never work!

Second, we’re going to use your server’s public IP address as the destination for incoming traffic to your NeoRouter VPN. If this is a dynamic IP address, you’ll need an FQDN that’s kept current by a service such as DynDNS.com.

Third, each administrator and user is going to need a username to access your NeoRouter VPN. You can use the same credentials to log in from multiple client machines, something you may or may not want to do. We’re going to set up credentials for one administrator as part of the install. You can add extra ones by adding entries with one of the following commands using the keyword admin or user. Don’t use any special characters in the username and password!

nrserver -adduser username password admin
nrserver -adduser username password user

Fourth, make up a very secure password to access your NeoRouter VPN. No special characters.

You’re done. Review your entries very carefully. If all is well, press Enter. If you blink, you may miss the completion of the install process. It’s that quick.

Fifth, after your NeoRouter VPN is installed, you can optionally go to the NeoRouter web site and register your new VPN by clicking Create Standalone Domain. Make up a name you can easily remember with no periods or spaces. You’ll be prompted for the IP address of your server in the second screen. FQDNs are NOT permitted.

When a VPN client attempts to login to your server, the server address is always checked against this NeoRouter database first before any attempt is made to resolve an IP address or FQDN using DNS. If no matching entry is found, it will register directly to your server using a DNS lookup of the FQDN. Whether to register your VPN is totally up to you. Logins obviously occur quicker using this registered VPN name, but logins won’t happen at all if your server’s dynamic IP address changes and you’ve hard-coded a different IP address into your registration at neorouter.com.

Setting Up a NeoRouter Client. As mentioned previously, there are NeoRouter clients available for almost every platform imaginable, except iPhones and iPads. Hopefully, they’re in the works. So Step #1 is to download whatever clients are appropriate to meet your requirements. Here’s the NeoRouter Download Link. Make sure you choose a client for the Free version of NeoRouter. And make sure it is a version 1.7 client! Obviously, the computing platform needs to match your client device. The clients can be installed in the traditional way with Windows machines, Macs, etc.

CentOS NeoRouter Client. As part of the installation above, we have automatically installed the NeoRouter client for your particular flavor of CentOS 6, 32-bit or 64-bit. In order to access resources on your NeoRouter server from other clients, you will need to activate the client on your server as well. This gets the server a private IP address in the 10.0.0.0 network.

To activate the client, type: nrclientcmd. You’ll be prompted for your Domain, Username, and Password. You can use the registered domain name from neorouter.com if you completed step #5. Or you can use the private IP address of your server. If your router supports hairpin NAT, you can use the public IP address or server’s FQDN, if you have one.

To exit from NeoRouter Explorer, type: quit. The NeoRouter client will continue to run so you can use the displayed private IP addresses to connect to any other online devices in your NeoRouter VPN. All traffic from connections to devices in the 10.0.0.0 network will flow through NeoRouter’s encrypted VPN tunnel. This includes inter-office SIP and IAX communications between Asterisk® endpoints.

Admin Tools for NeoRouter. Here are a few helpful commands for monitoring and managing your NeoRouter VPN.

Browser access to NeoRouter Configuration Explorer (requires user with Admin privileges)

Browser access to NeoRouter Network Explorer (user with Admin or User privileges)

To access your NeoRouter Linux client: nrclientcmd

To restart NeoRouter Linux client: /etc/rc.d/init.d/nrservice.sh restart

To restart NeoRouter Linux server: /etc/rc.d/init.d/nrserver.sh restart

To set domain: nrserver -setdomain YOUR-VPN-NAME domainpassword

For a list of client devices: nrserver -showcomputers

For a list of existing user accounts: nrserver -showusers

For the settings of your NeoRouter VPN: nrserver -showsettings

To add a user account: nrserver -adduser username password user

To add admin account: nrserver -adduser username password admin

Test VPN access: http://www.neorouter.com/checkport.php

For a complete list of commands: nrserver –help

To change client name from default pbx.local1:

  • Edit /etc/hosts
  • Edit /etc/sysconfig/network
  • Edit /etc/sysconfig/network-scripts/ifcfg-eth0
  • Edit /etc/asterisk/vm_general.inc
  • reboot

For the latest NeoRouter happenings, follow the NeoRouter blog on WordPress.com.

GPL2 License. The install-neorouter application is open source software licensed under GPL2. The NeoRouter Server and Client software is freeware but not open source. This installer has been specifically tailored for use on PBX in a Flash 2 servers, but it can easily be adjusted to work with virtually any Linux-based Asterisk system. If you make additions or changes, we hope you’ll share them on our forums for the benefit of the entire VoIP community. Enjoy!

Originally published: Wednesday, April 18, 2012



Need help with Asterisk? Visit the NEW 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…

  1. We’ve built a script to rename your PIAF2 server in all the right places. You can download it here. []

Introducing PPTP VPNs: The Travelin’ Man’s Best Friend

It’s been almost three years since we introduced VoIP Over VPN to securely interconnect Asterisk® servers. As LogMeIn® continues to squeeze the free Hamachi® VPN into oblivion, we’ll have a new, Really Free™ matrix VPN solution for you in coming weeks. This will let you interconnect up to 256 PBX in a Flash™ servers in minutes, not months, with no muss, no fuss, no fees, and no licensing worries. But today we want to begin VPN Month by turning our attention to those that need a virtual private network to connect back to a home office network or a home for that matter. This includes the traveling businessman or woman, the physician or lawyer with multiple remote offices, and any hub-and-spoke business such as a bank that has small branch offices that need to transparently link back to the mothership for network and communications services. The hidden beauty of PPTP VPNs is that all data (including phone calls) travels through an encrypted tunnel between the satellite office and home base. If you travel for a living and rely on other people’s WiFi networks for Internet access, a layer of network security will be a welcome addition.

Believe it or not, Microsoft introduced the Point-to-Point-Tunneling-Protocol (PPTP) with Windows 95. Back then we knew it as Dial-Up Networking. Suffice it to say that, in those days, PPTP was anything but secure. Unfortunately, the bad name kinda stuck. For the most part, the security issues have been addressed with the possible exception of man-in-the-middle attacks which are incredibly difficult to pull off unless you are a service provider or have access to the wiring closets of your employer. You can read the long history of PPTP VPNs on Wikipedia for more background. If you’re traveling to China or other democracy-challenged destinations, you probably shouldn’t rely upon PPTP for network security. If these security considerations aren’t applicable in your situation, keep reading because PPTP VPNs are incredibly useful and extremely easy to deploy for an extra layer of VoIP and network security in most countries that have severe wiretapping penalties in place.

PPTP VPNs also provide home-away-from-home transparency to home office network services. Simply stated, with a PPTP VPN, you get a private IP address on the home office LAN that lets you do almost anything you could have done sitting at a desk in the home office. There’s more good news. Fifteen years ago, we paid Cisco thousands of dollars for hardware boxes known as PPTP VPN Concentrators. You can still find some of them on eBay. For history buffs, a little company in California originally built these boxes. I think we paid about $3,000 for them. One year later Cisco bought the company and promptly doubled the price. Today, you can Do It For Free™ using your existing PIAF2 server platform. And, trust me, today’s 2-minute setup runs circles around the hoops we jumped through 15 years ago to install PPTP VPNs. Once deployed, they revolutionized mobile computing.

If you’re already running one or more PIAF2™ servers, then adding a PPTP VPN server to an existing system is a job for a Fifth Grader. Remember, you only need to do this on one server at your home base even if you have a dozen. The other good news is there are PPTP VPN clients for almost any platform you can name. Linux, Windows, Macs, Android, as well as iPhones, iPads, and iPod Touch devices all have free PPTP VPN clients that can be activated in less than a minute giving you instant, secure home base access.

Getting Started. We’re assuming you already have a PBX in a Flash 2 server set up behind a hardware-based firewall. If not, start there. Next, we’ll need to download and run the installer for your PPTP VPN Server. Just log into your server as root and issue the following commands:

wget http://incrediblepbx.com/install-pptp
chmod +x install-pptp
./install-pptp

UPDATE: For those of you still running a PBX in a Flash 1.7.x server under CentOS 5, we have a separate install script for you thanks to the great work of scurry7:

wget http://incrediblepbx.com/install-pptp-centos5
chmod +x install-pptp-centos5
./install-pptp-centos5

The Server Install: Five Easy Pieces. The installer will walk you through these five installation steps, but we’ll repeat them here so you have a ready reference down the road.

First, on your hardware-based firewall, map TCP port 1723 to the private IP address of your PIAF2 server. This tells the router to send all PPTP VPN traffic to your PIAF2 server when it hits your firewall. If you forget this step, your PPTP VPN will never work!

Second, you’re going to need a dedicated IP address on your private LAN to assign to the PPTP VPN server. Make sure it’s not an IP address from your router’s DHCP pool of addresses, and make sure it’s not one of the addresses from Step #3 below.

Third, you’re going to need two or more sequential IP addresses on your private LAN to assign to PPTP VPN clients that connect to your server. Remember, the PPTP design makes every remote client a node on your local area network so each client needs a private IP address on your LAN. Figure out how many client devices will be simultaneously connecting to your server and add one to it. Make sure the addresses you choose are in sequential order and not part of your router’s DHCP pool of addresses. Don’t use the address reserved for your PPTP server in Step #2 above. The address range should look something like this entry: 192.168.0.41-49. If you get the syntax wrong, guess what happens? If you screw it up, you can edit your localip and remoteip entries in /etc/pptpd.conf.

Fourth, each user is going to need a username to access your PPTP server. We’re going to set up credentials for one user as part of the install. You can add extra ones by adding entries to /etc/ppp/chap-secrets. For an extra layer of security, make the username as obscure as a password. Just don’t use any special characters. Upper and lowercase letters sprinkled with numbers are perfect. We recommend a length of at least 8 alphanumeric characters.

Fifth, make up an equally secure password to access your PPTP server. Same rules apply as in Step #4.

You’re done. Review your entries very carefully. If all is well, press Enter. If you blink, you may miss the completion of the install process. It’s that quick.

Configuring PPTP Client Devices. As we mentioned, there are available PPTP clients for Linux and Windows machines and Macs as well as Android and Apple smartphones and tablets. We’ve documented the steps for the various client setups on the PBX in a Flash Forum. Come visit! You’ll also discover some great tips from our resident gurus. We also would encourage you to post any questions that arise in your use of PPTP VPNs in that thread. You’ll get a quick and courteous response.

Secure VoIP Calling. The collateral benefit of implementing a PPTP VPN on your PIAF server is that all calls between remote extensions and home base can now be transmitted through a secure VPN tunnel. The only adjustment necessary using a SIP client on either an Android or Apple device is to replace the public server IP address with the server’s LAN IP address, and all of the communications traffic will flow through the VPN tunnel. The way we set up our Android phone with the Bria SIP client is to allocate an extension from the home office PIAF server to the SIP client and then enter the private IP address of the PIAF server in the Bria configuration. Then, when you’re at home base with WiFi, the client just works. And, when you’re on the road, just turn on the PPTP VPN, and Bria will register through the VPN tunnel using the exact same settings. It’s that easy, and it works great with WiFi or 3G/4G.

Checking for Connected Clients. If you get curious about who is logged into your PPTP server, here’s the command that’ll let you know: last | grep ppp.

GPL2 License. The install-pptp application is open source software licensed under GPL2. It has been specifically tailored for use on PBX in a Flash 2 (and now PIAF 1.7.x) servers, but it can easily be adjusted to work with virtually any Linux-based Asterisk system. If you make additions or changes, we hope you’ll share them on our forums for the benefit of the entire VoIP community. Enjoy!

What’s Next? For a more traditional client-server VPN which still relies upon a central server but uses a star topology to connect remote nodes, see this new Nerd Vittles article on the NeoRouter VPN.

Originally published: Monday, April 9, 2012



Need help with Asterisk? Visit the NEW 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…

The Perfect Threesome: iNum + VoIP.ms + Google Voice

We’ve got a terrific new VoIP development for you today especially for those who travel internationally. For several years, a VoIP company called VoxBone has been pushing hard to establish an International Number™ (iNum™) for every phone on the planet so that every telephone could call every other telephone at little or no cost. They’re not quite there, but two recent events will certainly hasten the implementation. The first was an announcement from VoIP.ms that they would provide a free iNum DID and free iNum calling to every one of their customers with a credit balance in their account. The second was last week’s announcement from Google that they, too, would support free iNum calling worldwide using any Google Voice account. Today, we’ll show you how to take advantage of these two developments to begin making free calls worldwide using your PBX in a Flash™ server, a WiFi-enabled smartphone, and an available WiFi connection. Basically, the plan is to use free iNum calling to get back to your PBX for dial tone and then use DISA for free Google Voice calling in the U.S. and Canada.

Until everyone has an iNum or Google opens up Google Voice outside North America, the hidden beauty of iNum for those of us who have both is the cost savings that can be achieved by phoning home with iNum from anywhere in the world for free. And, once the call hits your Asterisk® PBX, it’s incredibly simple to route the call to DISA, prompt for a password, and then place a call to anywhere in the U.S. or Canada at no cost with PIAF2™ and Google Voice.

This can be accomplished in several ways. First, you can download a SIP phone and use it in conjunction with your VoIP.ms account and a smartphone to make free iNum calls from any WiFi hotspot in the world. Bria is our favorite on both the iPhone/iPad and Android platforms. If $10 is too rich for your blood, there are some free alternatives: CSipSimple for Android and 3CXPhone for Android or iPhone. A second alternative is to use Google Voice or Gtalk to connect back to your PIAF2 server via iNum and then use DISA and your local trunks to place outbound calls. A final alternative is to take advantage of the numerous local numbers now available in many countries to phone home using iNum. The only cost of these calls is the cost associated with calling the local number. You’ll find a list of the local phone numbers to make these calls on the iNum web site or in the footnote to this article.1 So today we’ll show you how to set up your PIAF2 server to support free iNum calling. It’s a 15-minute project.

VoIP.ms Setup. To get started, if you’re not already a customer, register for a voip.ms account by filling out their registration form.

Once you submit the form, you’ll have to confirm your registration by clicking on the link that is emailed to you. Then you’re ready to login with your email address and the password you set up when you created your account. That’ll bring you to the Main Portal Page for your new voip.ms account.

You’ll need a positive balance in your VoIP.ms account in order to create your free iNum account so deposit some money using PayPal or a credit card by clicking Finances, Add Funds. The minimum deposit is $25 which can be used to make penny a minute calls in the U.S. and Canada or equally reasonable calls to any phone number in the world. We won’t be doing any of that today. For today, all of our calls will be free thanks to iNum and the generous support of VoIP.ms. But the nest egg will be there as a backup to your other PIAF2 VoIP providers which is an excellent idea anyway.

Like Vitelity, VoIP.ms lets you create subaccounts to compartmentalize your VoIP services. This makes it easy to use VoIP.ms on multiple PIAF2 servers or even standalone SIP telephones. It also provides added security by separating out account names and passwords for VoIP services from your main VoIP.ms portal account that let’s you manage your settings and VoIP funding, a very good idea. So let’s first set up an account to use with Asterisk just to show you how easy it is.

From the Main Portal Menu, click on Subaccounts, Create Subaccount. The Subaccount creation form will display. Fill it out so it looks something like this. Just click on the form below to enlarge it if you want a better view.

Once you’ve clicked the button to create the subaccount, it takes about a minute for voip.ms to activate it. Then click Main Menu, Portal Home. The bottom of the portal page will now show your subaccount.

Let’s create one more subaccount. We’ll use this one so that we can access VoIP.ms from a standard SIP app running on our iPhone or Android device. We can use the subaccount either to make outbound calls directly from VoIP.ms on a pay per minute basis, or we can use it to make free iNum calls. To create the subaccount, repeat the process above and fill in the blanks using your own credentials and a very secure password. Be sure to choose ATA device, IP Phone or Softphone for the Device Type. We always leave International Calls Disabled unless we really plan to make international calls. This will not affect your ability to make iNum calls, and it reduces your financial exposure in the event your subaccount is compromised. Never, ever use auto-replenishment from your credit card on a VoIP provider account from any provider.

Before we get too far along, let’s activate your new iNum DID. Click on DID Numbers, Order DID. When the DID Order Form displays, click on the iNum link to order your free iNum DID.

When the iNum DID order form displays, fill out the form by clicking on the POP location nearest to your server. Then, in the SIP/IAX Routing column, be sure to select the Subaccount we created previously rather than the default Main Account. Finally click the Click Here to Order button.

You’ll get a Confirmation display that shows your new iNum DID. Write it down! We’ve already set up the proper routing for your new iNum DID in the previous step so you can ignore the Managing Your DID message.

That completes the setup of your VoIP.ms account with your free iNum DID. Now let’s configure your PBX in a Flash server to support VoIP.ms and iNum. We’re assuming you already have a PBX in a Flash server configured with at least one Google Voice account activated. If not, stop here and complete that step using the PIAF2 tutorial and optionally the Incredible PBX 3 and Incredible Fax 2 tutorial.

Smartphone SIP Client Setup. We used the free cSipSimple Android app to set up a connection with our second subaccount at VoIP.ms using cSipSimple’s Basic Setup Wizard. Here are the entries required to gain connectivity:

Once your SIP client is connected to VoIP.ms through your smartphone, you can make free iNum calls using this dial syntax: 0118835100xxxxxxxx where xxxxxxxx is the last 8 digits of your iNum beginning with 0. As noted previously, you do NOT have to enable international calls on your VoIP.ms subaccount for these calls to go through.

PBX in a Flash iNum Setup. We’ll be using the FreePBX GUI to configure PBX in a Flash to support iNum. Using your browser, log into the IP address of your server: http://ipaddress/admin. When prompted for your username and password, use maint and whatever FreePBX password you assigned when your server was set up.

To simplify things, we’re going to set up 2 trunks: one for your VoIP.ms subaccount and another for iNum. Begin by choosing Trunks, Add SIP Trunk in the FreePBX GUI. For Trunk Name, use voipms. For Maximum Channels, choose 2. For the Dial Pattern, enter 1 | NXXNXXXXXX and, in Outgoing Settings for the PEER Details, enter the following using your subaccount name and password as well as the POP you chose for your subaccount:

canreinvite=yes
nat=yes
context=from-trunk
host=atlanta.voip.ms
secret=subacctpw
type=peer
username=137786_myinum
disallow=all
allow=ulaw
fromuser=137786_myinum
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

Leave all the fields for Incoming Settings blank. For the Registration String, the syntax is subacctname:subacctpw@atlanta.voip.ms:5060/8835100xxxxxxxx. Using our example and assuming you’re using the Atlanta POP, the entry would look like this where xxxxxxxx is your own 8-digit iNum beginning with 0:

137786_myinum:secretPassword21@atlanta.voip.ms:5060/8835100xxxxxxxx

Verify that your server got a successful registration with your VoIP.ms subaccount by clicking Tools, Asterisk Info, SIP Info.

Now click Setup, Trunks, Add Custom Trunk. For Trunk Name, use iNum. For Maximum Channels, choose 5. For Dial Pattern, use 0XXXXXX. including the period! For Custom Dial String, use SIP/0118835100$OUTNUM$@voipms.

Next, we need to create an Inbound Route. Use your full iNum DID number in the DID Number field, e.g. 8835100xxxxxxxx where xxxxxxxx is your personal iNum beginning with a 0. Activate CallerID Superfecta for the CID Lookup Source. And choose a Destination for the incoming iNum calls. This could be an extension, an IVR, or whatever else you’ve set up on your server. For now, route it to a working extension on your PBX so we can test it below. Then you can edit the inbound route and change it to any destination.

Finally, create an Outbound Route. Name the route OutiNum. For the Dial Pattern, use 0XXXXXX. with the trailing period. For the Trunk Sequence for Matched Routes, choose inum. After you save the trunk settings, move it to the top of your trunk listing in the right column of FreePBX. What this route does is allow you to call other iNum numbers (including your own) by simply dialing the last 8-digits of any iNum that begins with 8835100 or 0118835100. These 8 digits will ALWAYS begin with a 0.

Now let’s modify at least one of your existing Google Voice Outbound Routes so that you also can make iNUM calls with Google Voice by dialing from any extension using the full 8835100xxxxxxxx international number. Go to Outbound Routes and click on the name of one of your Google Voice trunks. Add the following new Dial Pattern and click Submit Changes: 8835100XXXXXXXX

Taking iNum for a Spin. To test things out, use a phone connected to an extension other than the one you chose to route incoming iNum calls to above. Dial the last 8 digits of your own iNum DID, and that extension should begin ringing. Answer the other extension and make sure you have audio in both directions. Next, dial your complete iNum DID beginning with 8835100. This should also cause the other extension to ring even though the call was initiated through your Google Voice trunk. If you’d like to get a Weather Report by Zip Code, we’ve set up an iNum for you to try. Just dial 09901997.
Enjoy!

Originally published: Monday, February 27, 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.

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…

  1. Local iNum Access Numbers include the following: []

Thumbs Up: A New Flash Drive Installer for PIAF2 + CentOS6

Original photo courtesy of Green House Co. Ltd.

With the advent of netbooks and the gradual disappearance of optical drives, it’s just a matter of time until USB thumb drives will be the only remaining physical installation method still available for most software. Look no further than Apple’s Lion OS if you don’t believe it. Of course, if Microsoft has its way, no installation of Linux will be available with some Windows 8 hardware… for your own safety, of course. We’ll leave that for the courts to sort out.

Since inception, one of the key goals of the PBX in a Flash™ project has been to provide an install option that works reliably with USB thumb drives. Thanks to the great work of bmore on the PIAF Forums, a USB Flash Drive installer was developed for PBX in a Flash 1.7.5.6.2. And today, we’re pleased to deliver a more flexible thumb drive installation method for 32-bit PIAF2™ installs running under CentOS™ 6.2. With this new thumb drive installer comes support for every current version of Asterisk® and FreePBX®.

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. And, 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_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. Here’s a quick overview of what happens next.

The PIAF2 installer then syncs the time on your server to NTP, installs the latest yum updates for CentOS 6.2, installs the versions of Asterisk and FreePBX you selected (HINT: Incredible PBX requires FreePBX 2.9) and some other utilities including WebMin, Festival and Flite text-to-speech support for Asterisk, and, of course, the Google Voice GUI which lets you configure PIAF2 to make free calls in the U.S. and Canada in a matter of seconds. Finally the PIAF2 installer patches your system to activate the IPtables firewall for both IPv4 and IPv6 as well as adding Fail2Ban monitoring for Asterisk, SSH, and your Apache web server.

As part of the install procedure, you also will be prompted to choose a version and master password for FreePBX and the other VoIP web utilities. Once your server reboots, you can log into the Linux CLI using your root password to obtain the IP address of your server. Then you can access the PIAF2 web GUI with a browser pointed to the same IP address. To access the FreePBX GUI, choose that icon from the Admin menu. Just click on the User button to get there. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose during the PIAF2 install. We’ll walk you through the install steps once we get your USB thumb drive set up.

PBX on a Flash

Here’s the 5-minute drill to get a USB thumb drive loaded with the latest and greatest 32-bit PIAF2 ISO. Once you get that far, follow the PIAF2 install steps outlined below to get your system up and running. In less than an hour, you’ll have a fully functioning, rock-solid reliable PBX that can meet all of your telephony requirements. And, remember, it’s free and always will be™.

Prerequisites. To get everything installed on your USB Flash Drive, you’ll obviously need at least a 1GB Flash Drive. HINT: 2GB flash drives may be cheaper! Next, you’ll need a Windows XP/Vista/7 computer on which to set up the thumb drive. On the Windows PC, you’ll need to download and install the latest, greatest version of ISO2USB from SourceForge. We recommend you also download and install the HP Formatting Utility for flash drives. Finally, you’ll need to download the 32-bit PIAF 2.0.6.2.1 ISO from SourceForge.

Creating USB Flash Drive. Step #1 is to partition and format your USB flash drive as a FAT32 device. Some flash drives are temperamental about the formatting step. We can’t recommend strongly enough using the HP Formatting Utility to make certain you get a reliable, properly formatted thumb drive! Also be careful that you are, in fact, formatting your thumb drive and not your Windows hard disk!

Step #2, once the device is properly formatted, run ISO2USB. You’ll get a screen that looks like what is shown above. Click on the … button to the right of DiskImage ISO and choose the PIAF2 ISO that you downloaded to your Desktop. Make certain that the destination device shown on the bottom line of the display is your USB flash drive. You do not want to accidentally trash your primary drive!

Here’s the tricky part to this. You need to know the drive names of the devices on the target machine where you ultimately will be using this thumb drive. Try these commands on your target machine using a Linux LIVE CD if you’re unsure: dmesg | grep logical AND dmesg | grep sectors. For most modern machines with IDE drives, the names will be sda, sdb, etc. For older machines, they may be hda, hdb. You’ll know if it doesn’t work. 🙂

The gotcha with CentOS 6.x is that, whenever you boot a machine using a USB flash drive with CentOS 6.x, the device names get switched for that boot only. The USB boot device becomes sda even if your hard disk on the system shows up as sda when it is running without a thumb drive. So… in the ISO2USB setup, change the Hard Disk Name to sdb, and change the USB Device Name to sda. For Foxconn hardware and AMD BIOS machines, use sdc instead of sdb. A few other systems use sdd. In all cases, use sda for the USB Device Name. And, as we noted, you’ll know quickly if you made the wrong choice. Just recreate the thumb drive using the next letter in the alphabet. 😉

Once you’ve double-checked your USB destination drive (HINT: the drive size is quite different), choose OK to begin. When the ISO install completes, don’t forget to Eject your USB flash drive before removing it from the Windows PC!

Using the USB Flash Installer. When using the new flash installer, remember that we need to boot your new machine from the thumb drive. On most newer Atom-based computers, you accomplish this by inserting the USB device, turning the machine on, and then pressing F12 during the boot sequence to choose the boot device. You’ll just have to watch the screen of your new computer to see if some other key is used to pull up the boot selection screen. If all else fails, you can adjust the boot sequence in the BIOS settings to boot first from the USB device. If you change your BIOS boot sequence, just remember to remove the device when the initial install of CentOS completes and the PIAF2 reboot sequence begins. If instead you again see the initial PIAF2 install screen warning you that your disk is about to be erased, then remove the thumb drive and reboot the machine once again.

PIAF Installation. Once you’ve booted with your PIAF2 thumb drive, you’ll be prompted to choose an installation method. For most users, simply pressing the Enter key will get things started. Choose a time zone when prompted and then enter a very secure root password for your new server. The installer then will load CentOS 6.2 onto your server. When complete, your server will reboot. Remove the thumb drive at this point, and you’ll be prompted to choose the version of Asterisk to install. See the discussion above for making a selection. If you see a Linux login prompt instead, it means sdb was the wrong device name for your server’s hard disk. Log in as root using the password you set up previously and issue the following commands to decipher the correct device name. Then rebuild your thumb drive using the correct device name and start again.

ls /dev/sd*
ls /dev/dd*

If all went well, after choosing the version of Asterisk to install, you’ll be prompted for a version of FreePBX and a master password for FreePBX. Make it very secure! We recommend FreePBX 2.9 if you plan to use Incredible PBX. Once you’ve made your choices, the PIAF2 installer will load Asterisk, FreePBX, and all the other PBX in a Flash components including Google Voice.

Once your server reboots, log into the Linux CLI using your root password and write down the IP address of your server from the status display.

Security Warning: Always, always, always run PBX in a Flash behind a secure, hardware-based firewall with no PBX in a Flash ports exposed to the Internet! After all, it’s your phone bill.

FreePBX Setup. Most of your life with PBX in a Flash will be spent using the FreePBX web GUI and your favorite browser. Just click on the image below to enlarge. To access the FreePBX GUI, point your browser at the IP address you wrote down. Read the RSS Feed in the PIAF GUI for late-breaking security alerts. Then click on the Users button which will toggle to the Admin menu. Click the FreePBX icon. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in completing the PIAF2 install.

To get a minimal system functioning, here’s the 5-minute drill. You’ll need to set up at least one extension with voicemail, configure a free Google Voice account for free calls in the U.S. and Canada, configure inbound and outbound routes to manage incoming and outgoing calls, and plug your maint password into CallerID Superfecta so that names arrive with your incoming calls. Once you add a phone with your extension credentials, you’re done.

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. Incredible PBX automatically randomizes all of the extension passwords for you.

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

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

Configuring Google Voice Trunk in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize PBX in a Flash to meet your specific needs. Click the 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.

CallerID Superfecta Setup. CallerID Superfecta needs to know your maint password in order to access the necessary modules to retrieve CallerID information for inbound calls. Just click Setup, CID Superfecta, and click on Default in the Scheme listings in the right column. Scroll down to the General Options section and insert your maint password in the Password field. You may also want to enable some of the other providers and adjust the order of the lookups to meet your local needs. Click Agree and Save once you have the settings adjusted.

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!

Adding Plain Old Phones. Before your new PBX will be of much use, you’re going to need something to make and receive calls, i.e. a telephone. For today, you’ve got several choices: a POTS phone, a softphone, or a SIP phone. Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device known as a Sipura SPA-3102. It’s under $70. Be sure you specify that you want an unlocked device, meaning it doesn’t force you to use a particular service provider. This device also supports connection of your PBX to a standard office or home phone line as well as a telephone.

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.

Where To Go From Here. We’ve barely scratched the surface of what you can do with your new PBX in a Flash system. If you’re new to all of this, then your next step probably should be the Nerd Vittles’ Incredible PBX 3.0 and Incredible Fax 2.0 tutorial. It’s a 5-minute addition. And, of course, all 50 Asterisk applications in Incredible PBX are free and always will be. Enjoy!

PBX on a Flash

Getting Your Own PIAF Thumb Drive. Some of you have asked about how to obtain your very own PIAF thumb drive. Well, it’s easy. Just make a contribution of $50 or more to the Nerd Vittles and PBX in a Flash projects by clicking the PayPal Donate button at the top of this page, and we’ll get one off to you pronto. And, thanks in advance for your support of freeware and open source projects!

Originally published: Monday, February 20, 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.

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…

Virtual Utopia: 1-Minute Asterisk Installs with PIAF2-OpenVZ

Thanks to the terrific work of Darrell Dillman, today we have a new OpenVZ template for PBX in a Flash 2™ to introduce. It features the very latest 64-bit CentOS™ 6.2 with Asterisk® 1.8 and FreePBX® 2.9. Using the new OpenVZ template, you can create unlimited virtual machines in about one minute per server! And you can boot your new virtual machines in about the same time. This new PIAF2-OpenVZ template includes the usual PIAF2™ Feature Set including Google Voice for free calling in the U.S. and Canada. Once installed, you can add Incredible PBX 3™ and Incredible Fax 2™ in a few clicks.

One of the real beauties of hosting your own Proxmox server is the flexibility it gives you to create and load a wide variety of virtual machines that each appear to users to be dedicated servers. This could include a dozen Asterisk servers, or it might be a mix of a dedicated Apache server, a Windows Server, an Asterisk server or two, as well as Joomla, Drupal, Zimbra, and many others from this list. The other obvious advantage is cost. Individual Asterisk servers can be had for $300 or less to host a small branch office. But a Proxmox server such as Dell's current offering can host a dozen dedicated systems for about $50 per server.

If you haven't heard of OpenVZ templates before, you've missed one of the real technological breakthroughs of the last decade. Rather than wading through the usual 30-60 minute ISO installation drill, with an OpenVZ template, all of the work is done for you. And it's quick. You can build a dozen PIAF2-Purple systems using an OpenVZ template in the time it takes to bake a pan of slice-and-bake cookies. And it's incredibly easy to then tie all of these systems together using either SIP or IAX trunks. Just follow our previous tutorial. For developers that want to try various Asterisk configurations before implementation and for trainers and others that want to host dedicated Asterisk systems for students, the OpenVZ platform is a perfect fit.

We'll start with the bad news before we get to the really exciting new Asterisk platform we're introducing today. All of the current Proxmox server software that supports OpenVZ virtual machines has a serious security flaw. For that reason, you would only want to run Proxmox behind a hardware-based firewall with no Internet port exposure. If you fail to heed this warning, you run the very real risk of having not only your Promox server compromised but also all of the virtual machines running on it. The good news is that this security flaw does not appear to affect the PBX in a Flash virtual machines which we are introducing today. Since no direct Internet access is required to have a perfectly functioning PIAF2 server, we still strongly recommend never exposing any server to direct Internet access. MORAL: No Internet port exposure for any of your servers means you can sleep like a baby. We recommend Proxmox 1.8 which is a free download from the Proxmox VE web site. To get optimum use from Proxmox, you'll also want a processor in your server that supports Kernel-based Virtual Machines (KVMs). This full virtualization solution requires an x86 processor containing virtualization extensions (Intel VT1 or AMD-V CPU2 is needed). HINT: Most of Dell's servers are not a problem. Regardless of the server you choose, make certain that you check the CPU specs before you buy. Also be aware that, in addition to Proxmox, there are many other OpenVZ platforms from which to choose.

Installing Proxmox. If you go the Dell route, you'll need an external USB CD or DVD drive to install Proxmox. Dell's optical drives aren't supported in the Proxmox boot image. So begin by downloading the Proxmox VE 1.8 ISO image and create your CD. Then boot your new server from the CD (by pressing F11 for the boot selection screen and choosing your USB external drive on Dell servers). Press Return to begin the install, agree to the license agreement, and click Next on the installer screen to begin. Choose your country, time zone, and keyboard layout. Next choose a secure password and provide a valid email address which is used to send you critical alerts from your Proxmox server. Finally, choose a hostname, specify a fixed IP address, netmask, gateway, and DNS servers and then press Next. Three minutes later, you'll have a new Proxmox server. Log in to your server as root and create a directory for your backups: mkdir /backup.

Enabling IPtables Firewall. IPtables works a little differently in the OpenVZ environment. It actually runs on the Proxmox host. There are just two steps to get it working. First, shut down every running VM on your Proxmox server using the web interface. When you're sure they're all stopped and while logged into your Proxmox server as root carefully enter the following two commands. Note that, because of the length, the sed command stretches to several lines which should be unraveled into a single line for the command to execute properly! Using a block-copy from a desktop machine to your SSH session is the safest method.

sed -i 's|ipt_REJECT ipt_tos ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length|ipt_REJECT ipt_tos ipt_TOS ipt_LOG ip_conntrack ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length ipt_state iptable_nat ip_nat_ftp|' /etc/vz/vz.conf

/etc/init.d/vz restart

Don't forget to set the system time on your server: dpkg-reconfigure tzdata

You're finished with the CLI at this point. Now you'll be able to configure IPtables within each of your OpenVZ virtual machines as explained below.

OpenVZ vs. ISO Images. One of the beauties of Proxmox is that it supports two different types of images to create virtual machines. An OpenVZ template is akin to a snapshot of an existing system while an ISO image is identical to the installer you normally would burn onto a CD in order to install a software application on your server. In short, you still have to go through the installation scenario when you create a virtual machine (KVM) from an ISO image. A virtual machine created from an OpenVZ image is ready for use the moment it is created. If you remember when instant-on televisions first were introduced, you'll also appreciate the difference in boot times between OpenVZ and KVM machines which boot an application installed from an ISO in much the same manner as you would experience on a standalone machine.

As with life, there's a dark cloud lurking behind every silver lining, and this is especially true in the Asterisk environment. OpenVZ containers rely upon a shared kernel, the one that actually boots the Proxmox server. KVM containers created from ISO images are self-contained with their own complete operating system and kernel. Thus, zaptel or dahdi cannot be loaded directly from an OpenVZ container. Instead one must rely upon a shared version of zaptel or dahdi loaded on the Proxmox server itself. As it turns out, this is no small feat and certainly not a task for mere mortals. Bottom Line: If you need conferencing or otherwise need a timing source for your Asterisk deployment, you will not want to use the OpenVZ approach at least for now. If you want to try it later, here is the message thread on the PBX in a Flash Forum. On the other hand, if you have more traditional VoIP requirements for your PBX, then the ease of installation and use of the OpenVZ image makes perfect sense. So let's start there assuming you understand the limitations.

Installing PIAF-OpenVZ Template. Using a web browser, download the new PIAF2-OpenVZ image to your Desktop. Once you have the OpenVZ image in hand, point your web browser to your Proxmox server: https://ipaddress. Accept the default certificate and login as root. You'll get a Welcome screen that looks something like what's shown above. Click on the Appliance Template option. In the Upload File section, choose the PIAF2-OpenVZ image on your Desktop and click Upload. Be patient. It's a big file. So go have a cup of coffee. You'll get a prompt when it's completed. You can also do this directly within the Proxmox server by logging in as root and issuing these commands to install the latest PIAF2-OpenVZ template:

cd /var/lib/vz/template/cache/
wget http://nerd.bz/zwU8zb
mv zwU8zb centos-6.2-purple1.8.8-piaf_2.0.6.2-5_amd64.tar.gz

Creating OpenVZ Virtual Machines. Once installed, you can build Asterisk 1.8.8.0 virtual machines to your heart's content... in about a minute apiece. Just choose Virtual Machine, Create to create a new virtual machine using the OpenVZ template you just uploaded. In the Configuration section, choose OpenVZ for the Type and pick your new OpenVZ template from the pulldown list. Fill in a Host Name, Disk Space maximum (in GB), Memory Allocation (1024 recommended), and a very secure (root) Password. The other defaults should be fine. In the Network section of the form, change to the Bridged Ethernet (veth) option which means the VM will obtain its IP address from your DHCP server. Make sure your DNS settings are correct for your LAN or use Google's DNS servers: 8.8.8.8 and 8.8.4.4. Here's how a typical OpenVZ creation form will look. Just click on the image to enlarge.

Once the image is created, start up the virtual machine, wait at least 60 seconds for the system to load, and then click on Open VNC Console. Asterisk will be loaded and running. Verify this on the status display. You can safely ignore the status messages pertaining to IPtables assuming iptables -nL shows that IPtables is functioning properly. You now have a PIAF-Purple base platform running Asterisk 1.8.8.0 and FreePBX 2.9. REMINDER: Be sure you always run both Proxmox AND your virtual machines behind a hardware-based firewall with no port exposure to the Internet!

Before you do anything else, log into your virtual machine using SSH and run passwd-master to secure the passwords for FreePBX GUI access to your system. Also be sure to set the correct time zone3 on your virtual machine:4

mv /etc/localtime /etc/localtime.bak
ln -s /usr/share/zoneinfo/America/Indianapolis /etc/localtime
date

Once you have secured your passwords, you're ready to set up Asterisk to make and receive calls. For the complete 5-minute tutorial, see this Nerd Vittles article. REMINDER: Once you have set up a Google Voice account, created an extension with a secure password, and created an inbound route for your incoming calls, don't forget to reload Asterisk from the CLI or Google Voice calling will fail: amportal restart.

Installing Incredible PBX and Incredible Fax. An alternative before configuring your system is to first install Incredible PBX and Incredible Fax. We recommend it. This gives you a turnkey, full-featured PBX with almost every Asterisk feature available on the planet. While logged into your server as root, issue this command to install Incredible PBX: install-incredpbx3. When the install completes, issue the following command to install Incredible Fax: install-incredfax2. Restart your virtual machine to complete the install.

Asterisk CLI Change. Finally, just a heads up that (once again) the Asterisk Dev Team appears to have changed the default behavior of the Asterisk CLI. With Asterisk 1.8, if you make outbound calls after loading the CLI, you will notice that call progress no longer appears in the CLI. To restore the standard behavior (since Moses), issue the following command: core set verbose 3. 🙄

Securing IPtables with a WhiteList. If you're running your virtual machines behind a hardware-based firewall with no Internet port exposure AND all of those on your private LAN are trusted, you can quit here. Otherwise, you need to lock down the IPtables firewall on your virtual machines to only permit access from trusted IP addresses. As delivered with Incredible PBX, all private IP addresses are authorized and a number of dangerous Internet services also are accessible. Here's how to fix it. Log into each VM and edit /etc/sysconfig/iptables: nano -w iptables. Change the section of entries that look like the following by inserting a # at the beginning of each entry. Once you've added the # characters, your entries should look like this:

#-A INPUT -p tcp -m tcp --dport 22 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 113 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 80 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 443 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 21 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 9001 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 9080 -j ACCEPT
#-A INPUT -p udp -m udp --dport 4569 -j ACCEPT
#-A INPUT -p udp -m udp --dport 5000:5082 -j ACCEPT
#-A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 4445 -j ACCEPT
#-A INPUT -p tcp -m tcp --dport 5038 -j ACCEPT

Now scroll down a bit in the file and find the entries that look like the following. NOTE: If you didn't install Incredible PBX, you'll need to manually add these entries:

-A INPUT -s 192.168.0.0/255.255.0.0 -j ACCEPT
-A INPUT -s 172.16.0.0/255.240.0.0 -j ACCEPT
-A INPUT -s 10.0.0.0/255.0.0.0 -j ACCEPT
-A INPUT -s 127.0.0.0/255.0.0.0 -j ACCEPT

Immediately below these private network entries, add additional entries using the actual IP addresses that are needed to administer your virtual machine. Also include the IP addresses of any remote telephones that are not covered by the private LAN entries above. Each entry should look like the following using the actual IP addresses needed:

-A INPUT -s 111.222.111.222 -j ACCEPT

IMPORTANT: Save your changes after making sure you've included an entry for the IP address from which you currently are accessing your server. Otherwise, you will lock yourself out of your server. Then restart IPtables: service iptables restart. Verify that the entries are the way you expect: iptables -nL. Now, with a browser, attempt to access the IP address of your virtual machine from an untrusted IP address, e.g. your cellphone. Then repeat from a trusted IP address. If all is well, you're done.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) and restart Asterisk: amportal restart.

Quirks, Gotchas, and Updates. The only quirk you will notice in the current virtual machines is that IP6tables may not be running. We're working on it. For the latest breaking news and updates about PIAF2-OpenVZ, visit this thread on the PIAF Forum. Don't forget your Valentine tomorrow. Enjoy!

Originally published: Monday, February 13, 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.


 

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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...

  1. Be very careful choosing Intel processors. Even some high-end processors do not support Intel Virtualization Technology. Here's the official list. []
  2. And here is a useful reference for AMD-compatible processors. The AMD WIKI provides the following list of AMD-V compatible processors: "AMD's x86 virtualization extension to the 64-bit x86 architecture is named AMD Virtualization, also known by the abbreviation AMD-V, and is sometimes referred to by the code name 'Pacifica'. AMD processors using Socket AM2, Socket S1, and Socket F include AMD Virtualization support. AMD Virtualization is also supported by release two (8200, 2200 and 1200 series) of the Opteron processors. The third generation (8300 and 2300 series of Opteron processors) will see an update in virtualization technology..." []
  3. Look in /usr/share/zoneinfo for correct time zone name for your closest city. []
  4. Getting the correct time in your VMs can be problematic with Proxmox. If you continually see the wrong time when you issue the date command after starting up your VMs, try this. Log into the Proxmox host and issue the following commands using the correct container number and your local time zone city for your virtual machine:

    vzctl stop 108
    vzctl set 108 --capability sys_time:on --save
    vzctl start 108
    vzctl enter 108
    mv /etc/localtime /etc/localtime.old
    ln -s /usr/share/zoneinfo/America/New_York /etc/localtime
    exit

    []

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.

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

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.

Incredible PBX 2.9. If you want all of the awesome Asterisk apps in one easy-to-install package, then Incredible PBX 2.9 is for you. Here's a link to the Nerd Vittles article explaining the 5-minute drill. Enjoy!

Originally published: Monday, December 26, 2011



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