PBX in a Flash Rolls Out New CentOS 5.7 Releases

We are pleased to announce the release of new 32-bit and 64-bit versions of PBX in a Flash. The new PIAF-17571 ISOs are available now for free download from SourceForge. In addition to an updated release of our new 64-bit CentOS 5.7 OpenVZ virtual machine template available on SourceForge, we now have a 32-bit Thumb Drive installer up on SourceForge as well.

So PBX in a Flash continues to bring you the best of all worlds: a hardware-based bare metal install using either our 32-bit or 64-bit ISOs to build a bootable CD-ROM installer, a 32-bit thumb drive installer for use with any 1GB USB flash drive to create PIAF systems on machines that lack an optical drive, or a 1-minute install of a virtual machine using our new 64-bit OpenVZ template. Nobody else provides this flexibility much less support for CentOS 5.7 as well as every current and experimental flavor of Asterisk. So why wait? The price is definitely right!

Today's step-by-step guide will walk you through installing PIAF-Purple with Asterisk 1.8.6.0 on a dedicated machine with a CD/DVD drive using the new CentOS 5.7 ISOs. Instructions for installation of the OpenVZ template on a virtual machine are provided in this updated Nerd Vittles article. Instructions for use of the flash drive installer are available in this updated Nerd Vittles article. As always, we recommend installation of any new PIAF server or virtual host behind a secure, hardware-based firewall (such as dLink's Gaming Router) with NO INTERNET PORT EXPOSURE to your PIAF box!

Atom-based PC Platform. For the least expensive hardware alternative, pick up an Atom-based PC, preferably not an EEE PC because of the network driver incompatibility with CentOS. The refurbished Revos work fine. Someone has actually tested them! They can easily support a business with dozens of phones.

PIAF ISO Setup. Once you have your hardware connected to a reliable Internet source, you'll need to choose the appropriate ISO for your hardware. If you have a CD-ROM or DVD drive on your server, we'd recommend the 32-bit PIAF 1.7.5.7.1 ISO. Just download it from SourceForge or one of the PIAF mirror sites, burn it to a CD, and then boot your server from the CD. If your server lacks a CD-ROM and DVD drive, then download the brand new 32-bit PIAF 1.7.5.7.1 Flash-Only ISO from SourceForge and copy it to a 1GB or larger thumb drive following the instructions in this Nerd Vittles tutorial. Then boot your server from the thumb drive.

PIAF Installation. Once you've booted the PIAF installer, you'll be prompted to choose an installation method. For most users, simply pressing the Enter key will get things started. Choose a keyboard and time zone when prompted and then enter a very secure root password for your new server. The installer then will load CentOS 5.7 onto your server. When complete, your server will reboot. Remove the CD or Flash Drive at this point, and you'll be prompted to choose the version of Asterisk to install. If you don't get the CD out in time, the install process will start from scratch. At the first prompt, just reboot after removing the CD and everything will be fine. We recommend PIAF-Purple. It loads Asterisk 1.8.6.0, the only current version of Asterisk with long-term support.

During the final phase of the install, you will be prompted to choose a master password for FreePBX® and the other VoIP web utilities. 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.

FreePBX Setup. Most of your life with PBX in a Flash will be spent using the FreePBX web GUI (click on image below to enlarge) and your favorite browser. 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. Any alerts older than September, 2011 already are included in current PIAF builds. Now 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 during the PIAF install.

Got That Pioneer Spirit? If you like living on the wild side, it's a simple process to upgrade the default FreePBX 2.8 install to FreePBX 2.9. Here's a 5-minute video that will walk you through the process. If you should get stumped, don't worry! Just visit this thread on the PIAF Forum. It'll cover everything you'll see in the video and more.

With either FreePBX 2.8 or 2.9, getting a minimal system operational is a 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. Now add a phone with your extension credentials and you're done.

Extension Setup. Let's start by setting up an extension. 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. PBX in a Flash does not!

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.

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. One terrific feature of CID Superfecta is the ability to test a phone number and see what results are returned by different services. It also tells you how long the various lookups are taking. Use this tool to narrow down the number of services you need and minimize the delay in answering inbound calls.

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

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 latest Incredible PBX 2.0 tutorial. It's a 5-minute addition that installs nearly 50 Asterisk applications that will keep you entertained for the rest of the year. If you'd prefer to do it yourself, that's okay, too. We'd also recommend you set up an alternate VoIP provider. You can't beat Vitelity, and they also happen to provide financial support to both Nerd Vittles and the PBX in a Flash projects. See the special pricing in the section below. Enjoy!

Originally published: Tuesday, September 27, 2011


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


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7 Responses to “PBX in a Flash Rolls Out New CentOS 5.7 Releases”

  1. Mike says:

    You might want to think about revising your recommendation for the Sipura SPA-3102. While it was a great piece of technology in its day, the Obihai OBi110 is less expensive, has more functionality, and is less prone to echo (the bane of the SPA-3102). Just sayin’…

  2. Raphael says:

    Great! I’m glad PBXinAFlash is always the best! I was wondering if someday you could post a complete guide for everysteps to upgrade for new versions. I can’t wait to test new things on a prod machine!
    Many thanks!

  3. Jim says:

    In the past, with installations of PIAF, you had to download incrediblepbx2.x and run it to get the IncrediblePBX features. You don’t mention it in this discussion of PIAF-17571. Does this now occur automatically with installation of the PIAF-17571 distribution?

    [WM: Incredible PBX 2 is still a separate add-on.]

  4. ty says:

    I tried to sign up for Vitelity via the PIAF special link. I enter the info and get my preqoalification message. When I click to go to Vitewlity to sign up, I get “400- Whoops! Page cannot be found” error. Is Vitelity still working with PIAF on it’s special offer?

    [WM: Special offer is still available. Vitelity was just moving servers. Try again, and thanks for the heads up.]

  5. Jim says:

    I feel pretty stupid in having asked that question after looking back at the article. It does specifically say to go the the Incredible PBX Tutorial in the article. I’d just done a quick search for wget and /root, didn’t find them in the article, and then posted my question.

    [WM: No problem at all.]

  6. Trousle Undrhil says:

    @Raphael: I’ve been asking for that too, but apparently, this project is no longer aiming to be the only one that doesn’t require constantly downloading and burning new ISOs.

    [WM: PBX in a Flash comes with update-source which will update CentOS to the latest and greatest. And FreePBX has a built-in upgrade utility to let you move up to FreePBX 2.9 or 2.10. Install scripts to upgrade Asterisk are on the PIAF Forum and can be modified to meet your specific requirements. Incredible PBX actually installs from an image of FreePBX and the Asterisk DB so it just isn’t possible to rerun the Incredible PBX installer without erasing all of your existing settings. It’s been that way since Day 1. Hope that clarifies the upgrade path and what’s available.]

  7. Dallas says:

    “preferably not an EEE PC because of the network driver incompatibility with CentOS.”

    Yes this is sad. :( I jumped at EEE Box because I thought it was good value. Pity about the network card – Fedora has no problem with it.

    [WM: Check out the PIAF Forum. There’s a thread on how to add the driver for the EEE box.]

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