The Ultimate VoIP Sandbox: PBX in a Flash 2 with CentOS 6

Today we're delighted to introduce the new PBX in a Flash 2™ beta built atop the latest release of CentOS® 6. Featuring superior scalability, improved performance, better resource management, and unmatched device support, PBX in a Flash 2.0.6.0 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. In addition to providing your choice of Asterisk 1.8 versions or Asterisk 10, PIAF2™ also gives you a choice of FreePBX® 2.8, 2.9, or 2.10. For those wanting to experiment, PIAF2 also provides direct access to Asterisk's menuconfig system to customize 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. Incredible PBX is not yet compatible!

NEWS FLASH: Looks like free Google Voice calling in the U.S. and Canada will be continued for 2012. See our Google+ post for details.

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 release. 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 DVD 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 CentOS6. When your system reboots, remove the DVD 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. Within a minute or so, your chosen Asterisk installer will load. In Phase 3 (the Config Module), you 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. 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.

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.

PIAF2 in the Cloud. We've been experimenting with several new (free) Cloud storage offerings. Because of the enormous size of the new ISO (1.79 GB), we've decided to host the PIAF2 beta ISO with two of these cloud providers in addition to some of our regular mirrors. This will let you judge the spectacular download performance of these new cloud offerings for yourself. Incidentally, you can sign up for your own free cloud storage at both sites. Our favorite is minus.com which offers 10GB of free cloud storage for life plus an extra gigabyte for you and for the PIAF Dev Team when you use our special signup link. You've got nothing to lose, and it helps the PBX in a Flash project as well. The other free offering is from one of our old favorites, PogoPlug. Just visit their site and grab your 5GB of free storage. Just a footnote that C|NET was offering 10GB of free PogoPlug storage last week, but you missed the window unless you're a PIAF Forum regular. HINT!

Creating a PIAF2 DVD. To get started, download the 1.79GB PIAF2 ISO. The MD5 checksum for the 32-bit ISO is 13d9302ef408feae726c2ca2b2c42a7c. The 64-bit MD5 is 3e7264e27099e35b631e7c7acca65c95. Here are the current download links.

Special Note: Upcoming Incredible PBX 2.9 only works with 32-bit ISO using PIAF-Purple (Asterisk 1.8) and FreePBX 2.9.

Minus Cloud (32-bit): Folder or ISO or Wget Link
Minus Cloud (64-bit): Folder or ISO or Wget Link
PogoPlug Cloud: Folder or 32-bit ISO
Google Cloud: Folder or 32-bit ISO
Vitelity: 32-bit ISO Download Link
SourceForge: Folder or 32-bit ISO or 64-bit ISO

Once you have the ISO image in hand, the next step is to burn the ISO image to a DVD. If you've never done it before, here's a tutorial that will show you how on either a Windows machine or a Mac.

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 using the Wget Link above. Once the download completes, don't forget to rename the ISO to pbxinaflash20601-i386.iso. The naming convention really matters with Proxmox! 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 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.

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. The PIAF Web GUI (shown above) is undergoing some upgrades to assure compatibility with the new PHP release. If you point a browser to the IP address of your server and see no icons or RSS Feed (as shown above), then apply the update patch below.

Update: Here's the two-line patch to address the PHP 5.3 change in behavior. The PHP opening tag syntax of <? is no longer allowed by default. To force PHP 5.3 to conform to the prior (permissible) syntax, log into your server as root and issue the following two commands:

sed -i 's|short_open_tag = Off|short_open_tag = On|' /etc/php.ini
service httpd restart

This restores the PIAF GUI to its former operation. Be sure to check the 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.

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. Because of the PHP 5.3 issues mentioned above, we recommend FreePBX 2.9 which already is compatible with this new release.

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 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 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!

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

Incredible PBX. Just a cautionary note that Incredible PBX 2 was not designed for use with CentOS 6. Don't try it! Give us a few weeks to make some needed adjustments, and we'll let you know when it is safe to have a go at it.

A Final Word About Beta Software. We take great pride in our software. Before it reaches beta stage, you can rest assured that it's not only been tested, but it actually works. That doesn't mean it's bug free. You can help enormously by reporting bugs. Just leave a comment here, or log into the PIAF Forum and let us know about any issues you encounter. We hope you're as thrilled with this new release as we are. Happy Thanksgiving!

Originally published: Monday, November 21, 2011




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


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


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


Some Recent Nerd Vittles Articles of Interest...

Be Sociable, Share!

2 Responses to “The Ultimate VoIP Sandbox: PBX in a Flash 2 with CentOS 6”

  1. ward says:

    NEWS FLASH: CentOS 6.1 was released today.

  2. Dunc says:

    Would love to start using this, but I’m already running a previous version of Incredible/Ultimate PBX. How does one go about upgrading? Why don’t you ever include those instructions? Or if one can’t upgrade, why don’t you include a backup and migration script or plan?

    [WM: Incredible Backup will let you migrate to a newer version. :wink: ]

Ringbinder theme by Themocracy