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Speech-to-Text Directory Assistance Comes to Asterisk

Monday, January 30, 2012 / 1 Comment on Speech-to-Text Directory Assistance Comes to Asterisk

Since the invention of the telephone, the most critical component has been the ability to match people's names to their phone numbers. Ma Bell did this with live operators (including my aunt) for many years. Then came automated lookups where you called a number for directory assistance and actually spoke the name of the person you wished to call. A computer then converted your speech to text and looked up the number in a database. Typically, the number was spoken back to the caller who then could place the call. Or "for a few cents more" the lookup service would actually place the call for you. For the learning impaired, this became a godsend when many metropolitan areas switched to 10-digit dialing.

Today, we'll show you how trivial it is to implement this yourself on any Asterisk® server using Google's new (free) speech-to-text web service which we introduced a few weeks ago. It's a 2-minute drill using PBX in a Flash™ with Incredible PBX™ and Google's Speech-to-Text web service. We'll be using a MySQL database to demonstrate the concept today, but it easily could be tweaked for use with any ODBC-compatible database. ODBC demos are included in Incredible PBX by dialing 222 or 223.

Many years ago we demonstrated how to quickly place calls to your friends by dialing the first three letters in their names with any phone connected to your Asterisk server using our freely available AsteriDex™ database. This has been incorporated into Incredible PBX by dialing 412 from phones connected to your PBX in a Flash server. Thanks to Google's new (free) speech-to-text web service, today we'll show you how trivial it is to tweak that application to replace 3-letter calling with spoken names of people to call with Asterisk. When you're finished, you'll be able to pick up any phone on your Asterisk server, dial 4-1-2, speak the name of an individual or company in your AsteriDex database, and have Asterisk automatically place the call for you.

Legal Disclaimer. What we're demonstrating today is how to use a publicly accessible web resource to respond to queries using a phone connected to your Asterisk server. We're assuming that Google has its legal bases covered and has a right to provide the public service they are offering. We are not vouching for Google or the services being offered in any way. By using our tutorial, YOU AGREE TO ASSUME ALL RISKS, LEGAL AND OTHERWISE, ASSOCIATED WITH USE OF THIS FREELY ACCESSIBLE WEB TOOL. NO WARRANTY EXPRESS OR IMPLIED IS BEING PROVIDED BY US INCLUDING ANY IMPLIED WARRANTY OF FITNESS FOR USE OR MERCHANTABILITY. You, of course, have an absolute right not to read our articles or implement our code if you have reservations of any kind or are unwilling to assume all risks associated with such use. Sorry for legalese, but it's the time in which we live I'm afraid. Plain English: "Don't Shoot the Messenger!"

Prerequisites. The easiest setup for this is a new PIAF2™ server. Once you have it running, install Incredible PBX 3 by logging into your server as root and issuing the command: install-incredpbx3. For complete instructions on Incredible PBX 3, here's the link to the Nerd Vittles tutorial. If you'd prefer not to go the Incredible route, then simply install AsteriDex 4 and then add the CallWho extension. Finally, you'll need to run the Wolfram Alpha for Asterisk one-click installer. This gets Google's speech-to-text components installed on your server. Now you're ready to tweak the CallWho app to use speech-to-text lookups through Google instead of 3-letter dialing.

Editing nv-callwho.php. Log in as root and edit nv-callwho.php in /var/lib/asterisk/agi-bin:

cd /var/lib/asterisk/agi-bin
nano -w nv-callwho.php

Press Ctrl-W. Search for where dialcode =. Replace it with where name =.

Now save the file with the change: Ctrl-X, Y, then press Enter.

If you'd prefer to use the latest, greatest (preconfigured) version, ignore the above and issue the following commands instead:

cd /var/lib/asterisk/agi-bin
wget http://nerd.bz/xnyJR3
tar zxvf callwho21.tgz
rm callwho21.tgz

Tweaking Your Custom Dialplan. While still logged in as root, you'll also need to edit extensions_custom.conf in /etc/asterisk:

cd /etc/asterisk
nano -w extensions_custom.conf

Press Ctrl-W. Search for 412. Now scroll down to the following lines:

exten => 412,9,Read(DIALCODE,beep,3)
exten => 412,10,NoOp(Name lookup: ${DIALCODE})
exten => 412,11,AGI(nv-callwho.php,${DIALCODE})

You'll want to replace those lines with the following 3 lines with no word wrap:

exten => 412,9(record),agi(speech-recog.agi,en-US)
exten => 412,10,Noop(= Script returned: ${status} , ${id} , ${confidence} , ${utterance} =)
exten => 412,11,AGI(nv-callwho.php,${utterance})

Finally, you'll want to adjust the spoken prompts in lines 412,6 and 412,8 to say something like this: "At the beep say the name of the person or company you wish to call. Then press the pound key."

Now save the file with your changes: Ctrl-X, Y, then press Enter

Finally, reload your Asterisk dialplan: asterisk -rx "dialplan reload"

Test Drive. To test things out, pick up a phone connected to your Asterisk server and dial 412. When prompted for the person or company to call, say "American Airlines" and then press the pound key.

Tweaking AsteriDex. You may need to make some minor adjustments to entries in your AsteriDex database to accommodate speech-to-text queries. For example, the sample entries include American Airlines and Delta Air Lines. Google translates the spoken words "air lines" as "airlines" so you'll need to modify the Delta entry, or it won't find a match. Similarly, there's a sample entry for "Emery Worldwide" but Google translates the spoken words as "emory worldwide." While capitalization doesn't matter, emory will not match emery. But, with a little tweaking, you'll have a very impressive, homegrown directory assistance service to impress all of your Friends and Family™. Enjoy!

Fuzzy Search Update. After we went to press, one of our favorite pundits on the PIAF Forum suggested that perhaps implementing fuzzy logic searches with MySQL would improve results, particularly with proper names. Great idea! It solved both the Delta Air Lines and Emery Worldwide lookup issues. And it turned out it was incredibly simple to implement. All that was required was replacing the existing $query command in nv-callwho.php (as explained above) with the following. This now has been incorporated into the preconfigured AGI script which is available for download above.

$query = "SELECT * FROM user1 where strcmp(soundex(name), soundex('$dialcode')) = 0";

For additional enhancements, see this thread on the PIAF Forum.

Asterisk TTS Bug. Be advised that certain newer releases of Asterisk have a text-to-speech bug which abnormally terminates TTS messages that have an embedded comma. If you have stored names in AsteriDex using Lastname, Firstname format, this may pose a problem. The simple solution is to either remove the commas or change them to periods. In the alternative, you can add the following line of code immediately below all existing lines of code beginning with $msg in nv-callwho.php. This, too, has been incorporated into the preconfigured AGI script above.

$msg = str_replace( ",", ".", $msg );

Originally published: Monday, January 30, 2012

Support Issues. With any application as sophisticated as Asterisk, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with Information, Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won't have to wait long for an answer to your question.

 

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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.
 

Introducing Incredible PBX 3.0 and Incredible Fax 2.0

Monday, January 23, 2012 / 1 Comment on Introducing Incredible PBX 3.0 and Incredible Fax 2.0

As Nerd Vittles begins its seventh year, a birthday bash is certainly in order. And today we have not one but two of Tom King's reworked masterpieces to introduce. The PIAF2™ introduction with CentOS 6.2™ and your choice of Asterisk® and FreePBX® versions has certainly brought its share of challenges. But, with the new year, we're finally comfortable recommending everyone make the switch. Almost everything is faster, more stable, and smoother with CentOS 6.2. Yes, the pain is worth the gain. But this new platform also meant significant rewrites of some of our VoIP workhorses, and today everything is finally ready for prime time.

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Incredible PBX 3.0™ brings literally dozens of turnkey Asterisk applications to your PIAF2 server, and the installation process is so simple a monkey could do it. And Incredible Fax 2.0™ delivers free faxing with HylaFax™ and AvantFax® in a setup process that's as simple as pressing the Enter key. When you're finished, you'll have one of the open source wonders of the world with free phone calls and faxing throughout the U.S. and Canada together with almost every Asterisk application ever developed. There's more good news. You don't have to be smarter than a fifth grader to get any of it installed and working reliably with Asterisk. In fact, all of the new installers now are rolled into the base PBX in a Flash 2.0™ installation. Just run two simple scripts, and presto. You're done!

The Incredible PBX 3 Inventory. For those that have never heard of The Incredible PBX, here's the current 3.0 feature set in addition to the base install of PBX in a Flash with the CentOS 6.2, Asterisk 1.8 or 10, FreePBX 2.9, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Incredible Fax, Hamachi VPN, and Mondo Backups are still just one command away and may be installed using the scripts included with Incredible PBX 3.

What began as a kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google's free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet and provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is still free, you'll never know you weren't connected via Ma Bell's overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play.

Just download the latest 32-bit or 64-bit PBX in a Flash 2.0.6.2 ISO from SourceForge, burn to then boot from the PIAF2 CD, choose the PIAF-Purple option to load Asterisk 1.8 or PIAF-Red to load Asterisk 10, and pick FreePBX 2.9 when prompted. Once the PIAF2 install is completed, just run the new Incredible PBX 3.0 installer: install-incredpbx3. In less than an hour, you'll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep you busy exploring for months.

Thanks to its Zero Internet Footprint™ design, Incredible PBX 3 remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.

Did we mention that all of this telephone goodness is still absolutely FREE!

Prerequisites. Here's what we recommend to get started properly:

Installing Incredible PBX 3.0. The installation process is simple and straight-forward. We're down to 3 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PIAF-Purple & FreePBX 2.9 using the PIAF2 ISO
2. Run Incredible PBX 3 installer
3. Configure Google Voice and a softphone or SIP phone

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash 2 installed. To use Incredible PBX 3, just install the latest 32-bit or 64-bit version of PBX in a Flash 2. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 6.2 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the PIAF-Purple or PIAF-Red payload. You'll then be prompted to choose your flavor of FreePBX. Choose FreePBX 2.9. Then set your time zone and set up a password for FreePBX access, and you're all set. As part of the install, yum now will automatically update your operating system with the latest updates for CentOS 6.2.

You can download the 32-bit PIAF2 from SourceForge. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve AND remove any USB flash drives! Press Ctrl-C to cancel.

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS 6.2, it will reboot. Remove the CD promptly. After the reboot, choose PIAF-Purple. In less than a minute, you'll be prompted for the FreePBX version you wish to install. Choose FreePBX 2.9 and fill in your choices for the remaining prompts. Then have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. You now have a PBX in a Flash base install. On a stand-alone machine, it takes 30-60 minutes. On a virtual machine, it takes about half that time. Log into your server with your root password and write down the server's IP address. You'll need it to access FreePBX with your browser.

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs, update-fixes, and passwd-master for you. So your system is relatively secure out of the box! See the Proxmox cautionary alert in the footnotes to this article!

Configuring Google Voice. You'll need a dedicated Google Voice account to support Incredible PBX 3. If you plan to use the inbound fax capabilities of Incredible Fax 2, then you'll want an additional Google Voice line that can be routed to the FAX miscellaneous destination using FreePBX. 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, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with Incredible PBX 3. Google Voice no longer is by invitation only so, 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. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...

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

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

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

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

Incredible PBX 3.0 Installation. Log into your server as root and issue the following commands to run The Incredible PBX 3 installer:

install-incredpbx3

When The Incredible PBX install begins, you'll be prompted for your FreePBX maint password. This is required to properly configure CallerID Superfecta for you. Your credentials never leave your server!

Now have another 15-minute cup of coffee. While you're waiting just make sure that you've heeded our advice and installed your server behind a hardware-based firewall. No ports need to be opened on your firewall to support Incredible PBX. Leave it that way!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

FINAL STEP. Once the Incredible PBX install completes, be sure to download the latest updates and patches for PBX in a Flash and Incredible. Just issue the following commands:

update-programs
update-fixes

Logging in to FreePBX 2.9. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Users tab. It will change to Admin. Now click the FreePBX button. When prompted for a username, it's maint. When prompted for the password, it's whatever you set up as your maint password when you installed Incredible PBX 3. If you forget it, you can always reset it by logging into your server as root and running passwd-master.

Configuring Google Voice Trunks in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize your Incredible PBX to meet your specific needs. If you plan to use Google Voice, here's how to quickly configure one or more Google Voice trunks within FreePBX. After logging into FreePBX with your browser, 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. You can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill.

While you're still in FreePBX, choose Setup, Extensions, and click on the 701 extension. Write down your extension password which you'll need to configure a phone in a minute.

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.

Incredible Fax 2 Installation. If you want the added convenience of having your Incredible PBX double as a free fax machine, run install-incredfax2 after the Incredible PBX 3 install completes. Plug in your email address for delivery of incoming faxes and enter your home area code when prompted. For every other prompt, just press the Enter key. If you'd like to also add the optional OCR utility, just choose it when prompted. For complete documentation, see this Nerd Vittles article. Don't forget that a REBOOT OF YOUR SERVER is requiredwhen the install is finished, or faxing won't work! Then log in through the PIAF GUI using maint:password. Be sure to change your password!

Also be sure to set up a second, dedicated Google Voice number if you want support for inbound faxing. Once the Google Voice credentials are configured in FreePBX for the additional Google Voice line, simply add an Inbound Route for this DID to point to the FAX Misc. Destination that comes preconfigured with Incredible PBX 3. Substitute your 10-digit Google Voice number for the DID number shown below. Save your entries and reload FreePBX.

Extension Password Discovery. If you're too lazy to look up your extension 701 password using the FreePBX GUI, you can log into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone or color videophone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

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 above. 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 your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

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 with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password. We're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX and uncheck the Disable Trunk checkbox. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.

We've also included Trunk configurations for a dozen of our favorite hosting providers to get you started. You can sign up for service with any of them, insert your credentials in the existing trunk, uncheck the Disable Trunk checkbox, and then adjust your outbound route and add an inbound route for your new DID (if you get one).

Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it's merely a greeting. To those "in the know," it's actually an AutoAttendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It's quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It's your phone bill. 😉

Configuring Email. You're going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Setup, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Setup, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. We also recommend you follow @NerdUno on Twitter. We'll keep you entertained and provide immediate notification of security problems that we hear about. Be safe!

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.

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PIAF Support Conference Bridge (Conf#: 1061)
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)

PBX in a Flash SQLite Registry. Last, but not least, we want to introduce you to the new PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here's the complete list of options:

  • A - Write the contents of the registry to registry.txt
  • B - PBX in a Flash install details
  • C - Extra programs install details
  • D - Update-fixes status and details
  • E - RPM install details
  • F - FreePBX modules install details
  • G - Incredible PBX install details
  • Q - Quit this program

And here's a sample from an install we recently completed.

Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. To Philippe Sultan and the rest of the Asterisk development team, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer, Tony Lewis, and the rest of the FreePBX development team, thanks for FreePBX 2.9 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made CentOS 6 and Asterisk 1.8 and 10 sit up and bark. No one will quite understand what an endeavor that is until they try it themselves. You won't find another CentOS 6 implementation of Asterisk, and Tom has made it look incredibly easy. It wasn't! And, last but not least, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.

Additional Goodies. Be sure to log into your server as root and look through all of the free scripts that are included. Just type: help-pbx.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

Originally published: Monday, January 23, 2012

Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with Information, Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won't have to wait long for an answer to your question.


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

 

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.
 

  1. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB CD/DVD drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. You also can run Incredible PBX 3 on a virtual machine such as the free Proxmox server. A security vulnerability has been reported in the Proxmox browser so be sure to run your server behind a secure, hardware-based firewall with no port exposure to the actual Proxmox server from the Internet. [↩]

Introducing PBX in a Flash 2 with CentOS 6.2

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Another terrific option (if you hurry) is this refurbished Dell GX620 for $79.99. These won't last long.

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


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...

Introducing PIAF2 and Incredible PBX 2.9 with CentOS 6.1

Thursday, December 15, 2011


We're pleased to introduce the latest and greatest PBX in a Flash™ 2.0.6.1 featuring CentOS® 6.11 and the brand-new Incredible PBX™ 2.9 with an incomparable VoIP feature set. PIAF2™ provides turnkey installs of Asterisk® 1.8 or 2.0 with your choice of FreePBX® 2.8, 2.9, or 2.10. And, for those choosing to install Incredible PBX 2.9, it's been engineered to work flawlessly with the 32-bit version of PIAF2 using Asterisk 1.8 and FreePBX 2.9. For the ultimate in performance, a 64-bit version of PIAF2 is also available; however, because of its size, a DVD is required to burn the ISO. And, as noted, it is not compatible with Incredible PBX 2.9.

12/17 Update: Shortly after PIAF2 hit the street, Digium released Asterisk 1.8.8.0 and the first non-beta version of Asterisk 10. New 32-bit and 64-bit PIAF 2.0.6.1.2 ISOs will be available on SourceForge today that incorporate these new builds. In addition, a CentOS 6 video quirk has been identified on some Atom hardware. So the new ISOs include an install option to disable the problematic video testing by kicking off the install with one of the following commands instead of merely pressing the Enter key: ks-nomode, ksraid-nomode, or kslvm-nomode. You'll know if you have the problem if your server locks up. 😉 Finally, because there now are multiple stable versions of Asterisk, we have added the option to selectively choose a version of Asterisk to install. Instead of picking PIAF-Purple or PIAF-Red, you can drop down to the Linux command prompt, log in as root, and issue a command using the following syntax: piafdl -p beta_1872_purple.

Photo courtesy of mashable.com

Free Google Voice calling in the U.S. and Canada has been extended for calendar year 2012 and now can be configured using the simple FreePBX 2.9 GUI. And you can use it with or without Incredible PBX. Set up one or many Google Voice connections in less than 10 seconds per line. With Incredible PBX, we've also included Andrew Nagy's terrific EndPoint Manager that lets you configure dozens of SIP phones with the click of a button. You'll also find Kennonsoft's terrific new PBX in a Flash UI with HTML5 and CSS3 support for the latest Firefox, Chrome, and IE browsers. And, of course, you still get almost every Asterisk application on the planet preconfigured and ready to use.

With PIAF2, the installation process has been streamlined considerably. At the outset, you will be prompted for some basic information and a root password. Once the CentOS 6.1 install completes and you remove the CD/DVD during the server reboot, you will be prompted for whether you wish to tailor Asterisk using menuconfig, your time zone, the version of FreePBX you wish to install, and your master password for FreePBX access. Once you've answered these few questions, you can kick off the PIAF2 install and walk away. Depending upon the performance of your server, come back in 15-30 minutes. While it's not the quickest install on the planet, it will always be the most current because PIAF2 always loads the latest patches to CentOS as well as Asterisk and FreePBX. In other words, it's worth the wait to know you're installing a secure and up-to-date system. And, as your high school girlfriend probably taught you, faster is not always better.

The Incredible PBX 2.9 Inventory. For those that have never heard of The Incredible PBX, here's the current 2.9 feature set in addition to the base install of PBX in a Flash with the CentOS 6.1, Asterisk 1.8, FreePBX 2.9, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Faxing, Hamachi VPN, and Mondo Backups are still just one command away and may be installed using the scripts included with base Incredible PBX 2.9 installation.

Update: Incredible Fax is not yet compatible with PIAF2, but we're working on it.

What began as a kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google's free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet and provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is still free, you'll never know you weren't connected via Ma Bell's overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play.

Just download the latest 32-bit PBX in a Flash 2.0.6.1 ISO from SourceForge, burn to then boot from the PIAF2 CD, choose the PIAF-Purple option to load Asterisk 1.8, and pick FreePBX 2.9 when prompted. Once the PIAF2 install is completed, just run the new Incredible PBX 2.9 installer. In less than an hour, you'll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep you busy exploring for months.

Thanks to its Zero Internet Footprint™ design, Incredible PBX 2.9 remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.

Did we mention that all of this telephone goodness is still absolutely FREE!

Prerequisites. Here's what we recommend to get started properly:

Installing Incredible PBX 2.9. The installation process is simple and straight-forward. We're down to 3 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PIAF-Purple & FreePBX 2.9 using 32-bit PIAF2 ISO
2. Download & run Incredible PBX 2.9 installer
3. Configure Google Voice and a softphone or SIP telephone

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash 2.0 installed. To use Incredible PBX 2.9, just install the latest 32-bit version of PBX in a Flash 2.0. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 6.1 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the PIAF-Purple payload to get the latest Asterisk 1.8. You'll then be prompted to choose your flavor of FreePBX. Choose FreePBX 2.9. Then set your time zone and set up a password for FreePBX access, and you're all set. As part of the install, yum now will automatically update your operating system to CentOS 6.2 minus the 6.2 kernel.

You can download the 32-bit PIAF2 from SourceForge. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve AND remove any USB flash drives! Press Ctrl-C to cancel.

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS 6.1, it will reboot. Remove the CD promptly. After the reboot, choose PIAF-Purple. In less than a minute, you'll be prompted for the FreePBX version you wish to install. Choose 2.9 and fill in your choices for the remaining prompts. Then have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. You now have a PBX in a Flash base install. On a stand-alone machine, it takes 30-60 minutes. On a virtual machine, it takes about half that time. Log into your server with your root password and write down the server's IP address. You'll need it to access FreePBX with your browser.

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs, update-fixes, and passwd-master for you. So your system is relatively secure out of the box! See the Proxmox cautionary alert in the footnotes to this article!

Configuring Google Voice. You'll need a dedicated Google Voice account to support Incredible PBX 2.9. If you plan to use the inbound fax capabilities of Incredible PBX 2.9, then you'll want an additional Google Voice line that can be routed to the FAX miscellaneous destination using FreePBX. 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, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with The Incredible PBX. Google Voice no longer is by invitation only so, 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. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...

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

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

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

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

Incredible PBX 2.9 Installation. Log into your server as root and issue the following commands to download and run The Incredible PBX installer:

cd /root
wget http://incrediblepbx.com/incrediblepbx29.x
chmod +x incrediblepbx29.x
./incrediblepbx29.x

When The Incredible PBX install begins, you'll be prompted for your FreePBX maint password. This is required to properly configure CallerID Superfecta for you. Your credentials never leave your server!

Now have another 15-minute cup of coffee, and consider a modest donation to Nerd Vittles... for all of our hard work. 😉 You'll find a link at the top of the page. While you're waiting just make sure that you've heeded our advice and installed your server behind a hardware-based firewall. No ports need to be opened on your firewall to support Incredible PBX. Leave it that way!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

Logging in to FreePBX 2.9. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Users tab. It will change to Admin. Now click the FreePBX button. When prompted for a username, it's maint. When prompted for the password, it's whatever you set up as your maint password when you installed Incredible PBX 2.9. If you forget it, you can always reset it by logging into your server as root and running passwd-master.

Configuring Google Voice Trunks in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize your Incredible PBX to meet your specific needs. If you plan to use Google Voice, here's how to quickly configure one or more Google Voice trunks within FreePBX. After logging into FreePBX with your browser, 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. You can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill.

While you're still in FreePBX, choose Setup, Extensions, and click on the 701 extension. Write down your extension password which you'll need to configure a phone in a minute.

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.

Incredible Fax Installation. If you want the added convenience of having your Incredible PBX double as a free fax machine, run /root/incrediblefax.sh shell script when the Incredible PBX install completes. Plug in your email address for delivery of incoming faxes and enter your home area code when prompted. For every other prompt, just press the Enter key. For complete documentation, see this Nerd Vittles article. Don't forget to REBOOT YOUR SERVER when the install is finished, or faxing won't work!

Also be sure to set up a second, dedicated Google Voice number if you want support for inbound faxing. Once the Google Voice credentials are configured in FreePBX for the additional Google Voice line, simply add an Inbound Route for this DID to point to the FAX misc. destination that comes preconfigured with Incredible PBX 2.9. Just substitute your 10-digit Google Voice number for the DID number shown below. Save your entries and reload FreePBX.

Extension Password Discovery. If you're too lazy to look up your extension 701 password using the FreePBX GUI, you can log into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone or color videophone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

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 above. 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 your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

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 with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password. We're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX and uncheck the Disable Trunk checkbox. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.

We've also included Trunk configurations for a dozen of our favorite hosting providers to get you started. You can sign up for service with any of them, insert your credentials in the existing trunk, uncheck the Disable Trunk checkbox, and then adjust your outbound route and add an inbound route for your new DID (if you get one).

Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it's merely a greeting. To those "in the know," it's actually an AutoAttendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It's quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It's your phone bill. 😉

Configuring Email. You're going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Setup, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Setup, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. We also recommend you follow @NerdUno on Twitter. We'll keep you entertained and provide immediate notification of security problems that we hear about. Be safe!

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)

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PIAF Support Conference Bridge (Conf#: 1061)
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)

PBX in a Flash SQLite Registry. Last, but not least, we want to introduce you to the new PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here's the complete list of options:

  • A - Write the contents of the registry to registry.txt
  • B - PBX in a Flash install details
  • C - Extra programs install details
  • D - Update-fixes status and details
  • E - RPM install details
  • F - FreePBX modules install details
  • G - Incredible PBX install details
  • Q - Quit this program

And here's a sample from an install we recently completed.

Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. To Philippe Sultan and the rest of the Asterisk development team, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer, Tony Lewis, and the rest of the FreePBX development team, thanks for FreePBX 2.9 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made CentOS 6 and Asterisk 1.8 and 10 sit up and bark. No one will quite understand what an endeavor that is until they try it themselves. You won't find another CentOS 6 implementation of Asterisk, and Tom has made it look incredibly easy. It wasn't! In fact, when CentOS released 6.1 this week, Tom actually shifted gears (again) and rebuilt PIAF2 (in a couple of days) to take advantage of CentOS 6.1. And, last but not least, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.

Additional Goodies. Be sure to log into your server as root and look through the scripts added in the /root and /root/nv folders. You'll find all sorts of goodies to keep you busy. There's an all-new incrediblefax.sh script that painlessly installs and configures HylaFax and AvantFax for state-of-the-art faxing. The 32-bit install-cepstral script does just what it says. With Allison's Cepstral voice, you'll have the best TTS implementation for Asterisk available. ipscan is a little shell script that will tell you every working IP device on your LAN. trunks.sh tells you all of the Asterisk trunks configured on your system. purgeCIDcache.sh will clean out the CallerID cache in the Asterisk database. convert2gsm.sh shows you how to convert a .wav file to .gsm. munin.pbx will install Munin on your system while awstats.pbx installs AWstats. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. All the other scripts and apps in /root/nv already have been installed for you so don't install them again.

If you've heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud as well. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It's perfect for off-site backups and is included as one of the backup options in the PBX in a Flash backup utilities.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

Originally published: Thursday, December 15, 2011

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Safely Interconnecting Asterisk Servers for Free Calling

Adding Skype to The Incredible PBX

Adding Incredible Fax to The Incredible PBX

Adding Incredible Backup... and Restore to The Incredible PBX

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with Information, Patches and Bug Fixes for Incredible PBX 2.9. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won't have to wait long for an answer to your question.


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

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

 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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

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

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

Some Recent Nerd Vittles Articles of Interest...

  1. As part of the yum update process, you'll actually end up with CentOS 6.2 minus the 6.2 kernel. [↩]
  2. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB CD/DVD drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. You also can run Incredible PBX 2.9 on a virtual machine such as the free Proxmox server. A security vulnerability has been reported in the Proxmox browser so be sure to run your server behind a secure, hardware-based firewall with no port exposure to the actual Proxmox server from the Internet. [↩]

7 Steps to Skytopia: Pain-Free Calls with Skype and Asterisk

Tuesday, November 1, 2011 / 5 Comments on 7 Steps to Skytopia: Pain-Free Calls with Skype and Asterisk

As you probably know, Digium® announced that Skype for Asterisk® would not be available for sale or activation after July 26, 2011. Here we are in November. So what to do? If you're looking for a commercial solution, you're S.O.L. But, if you have a non-commercial PBX for personal use1, then keep reading. We'll walk you through, step-by-step, getting Skype integrated into your PBX in a Flash or Incredible PBX environment. It's easy, but it's a manual process. If you follow the steps below in order, you'll be up and running in about 15 minutes.

Prerequisites. For today's project, we're assuming you have an existing Incredible PBX server running CentOS 5.7. If not, here's our tutorial to get you up and running quickly. You'll also need a keyboard, mouse, and monitor. We strongly recommend a dedicated server such as an Atom-based PC. If you're using a virtual machine, then you'll need a sound card alternative. Try this: /sbin/modprobe snd-dummy.

UPDATE: We've revised this article a bit to accommodate PIAF2 with CentOS 6.2 and Incredible PBX 3. Keep in mind that Skype is a 32-bit application so we strongly recommend a 32-bit platform if reliability matters to you.

Step 1. For inbound Skype calling to work with other implementations including generic PBX in a Flash systems, you'll need to create a SIP URI for your Asterisk server: mothership@127.0.0.1. You do NOT need to expose the SIP port(s) of your Asterisk server to the Internet, and we strongly recommend that you don't! We've previously explained how to set up a SIP URI in this article. The Incredible PBX includes this SIP URI functionality out of the box.

Step 2. You'll also need Java 1.5. To see if it's included in your distribution, issue the following command: rpm -q jdk. If your particular Asterisk distribution doesn't have JAVA 1.5 or higher installed (rpm -q jdk), here's how to do it. Go to the Oracle Technology Network, sign up for a free Oracle web account and log in. While still logged in, accept the binary code license agreement, and click on this link to download jdk-6u12-linux-i586-rpm.bin. Then copy the file to /root on your Asterisk server. Make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin) and then run it. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Whew!

Step 3. You'll also obviously need a dedicated Skype account for your Asterisk server. If you don't have one to spare, download the Skype software for your Mac or Windows PC, and sign up for a free account. You can try out your account by calling our demo hotline: nerdvittles. Get this working on your Mac or PC before proceeding! Then be sure you log out and disable automatic logins on reboot, or you'll have a problem down the road with two machines trying to log in to a single Skype account.

Step 4. Now we're ready to install the remaining software components that your server will need to access Skype. Log into your Asterisk server as root and issue the following commands.

cd /root
mkdir skype
cd skype
wget http://download.skype.com/linux/skype_static-2.1.0.47.tar.bz2
tar jxvf skype_static*
yum -y install xorg-x11-server-Xvfb
yum -y install qt4
yum -y install xterm
yum -y install libXScrnSaver.i386 < == use this for CentOS 5.x #yum -y install libXScrnSaver <== use this for CentOS 6.x wget http://incrediblepbx.com/siptosis.tgz cd .. wget http://incrediblepbx.com/skype-start chmod +x skype-start cp skype-start skype/. cd / tar zxvf /root/skype/siptosis.tgz cd /root/skype

If you'd prefer to avoid all the typing, you can issue the following commands to download a script that will do all the heavy lifting for you. This is for CentOS 5.x systems only:

cd /root
wget http://incrediblepbx.com/skype-setup
chmod +x skype-setup
./skype-setup

For PIAF2 systems running CentOS 6.x, use this instead:

cd /root
wget http://incrediblepbx.com/skype-setup2
chmod +x skype-setup2
./skype-setup2

Step 5. Now there are a few steps to manually configure the software components so that the entire Skype startup process can be automated when your server boots in the future. To begin, you'll need to fire up X-Windows which puts your server in graphics mode. This is the only mode that Skype understands. While logged into your server as root, issue the following command: xinit

NOTE: If xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for CentOS 5.x systems:

Section "ServerLayout"
Identifier "X.org Configured"
Screen 0 "Screen0" 0 0
EndSection

Section "Device"
Identifier "Card0"
Driver "vesa"
EndSection

Section "Screen"
Identifier "Screen0"
Device "Card0"
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
EndSection

For PIAF2 users, some have reported issues on Atom machines with seeing a display at all after xinit loads. If this happens to you, don't panic. Simply log into your server from a PC or MAC using SSH. Then run: vncserver :1. Set a password for VNC, and then use a VNC client on your PC or Mac to access VNC at the IP address of your server on display port 1. Now you can continue with Step 6, below.

Step 6. Now we're ready to start up Skype, and get it properly configured. There are two important requirements. First, we want to make sure your credentials are saved for automatic login in the future. And second, we want to configure Skype to run in a minimized state each time it restarts. To begin, click in the white graphics window on your screen using your mouse and issue these commands:

cd /root/skype/skype_static-2.1.0.47
./skype

Click on the Accept button to accept the Skype license agreement. Once Skype loads, enter your Skype Name and Password. Before clicking on Sign In, be sure to check the Automatic Sign In box so that you'll be logged in automatically in the future. Once you're logged in, click on the blue S in the lower left corner of the window to access the Skype Main Menu. Then click Options. When the General tab displays, check the box which says Start Skype minimised in the system tray. Then click the Apply button. To test things out, click on the Sound Device tab and then Make a Test Call. Once you're sure everything is working, click the Close button. Now click on the blue S again and click Quit to shut down Skype.

Step 7. Now we're ready to integrate Skype into the SipToSis middleware so that Asterisk can communicate with Skype. Issue the following commands to start Skype in background mode and then start SipToSis. Be sure to write down the PID for Skype in case we need to kill the app if something goes wrong.

./skype &
cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype. Before clicking Yes, be sure to click the Checkbox to Remember This Selection for future connections! When you click Yes, you'll see the SipToSis CLI indicating that it's waiting for a Skype call.

If you've installed this on an Incredible PBX, Skype should now be functional. From another Skype account, just call the Skype Name that you used to set this up, and your Asterisk extensions should start ringing. To test outbound Skype calling, use an X-Lite softphone connected to an extension on your Asterisk server and dial *echo123 to access Skype's call testing service or *nerdvittles to access our demo.

All that remains is to configure your server to automatically start Skype and SipToSis whenever your system is restarted. Here's how. Press Ctrl-Alt-F2 to get a new login prompt on your server. Log in as root and issue the following command:

echo "/root/skype-start" >> /etc/rc.d/rc.local

Now reboot your server and make sure everything is working.

Navigation Tips. Here are a few navigation tips for managing your Asterisk console on CentOS systems once Skype has been installed:

1. Ctrl-Alt-F2 gets you a new login prompt for your server

2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To find the Skype PID: pidof skype. To kill Skype: kill pid#. To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_Linux

3. Ctrl-Alt-F9 gets you to the Asterisk CLI.

Setting Up Speed Dials for Skype Friends. One of the wrinkles with Skype is that Skype uses names for its users rather than numbers. If you don't have a SIP URI-capable softphone, there's still an easy way to place calls to your Skype friends using FreePBX®. Just add a Speed Dial number to your FreePBX dialplan. Choose Extension, then select the Custom type, provide an Extension Number which is the Speed Dial number (this could actually spell your friend's name using a TouchTone phone), enter a Display Name for your friend, and add an optional SIP Alias. Then insert the following in the dial field replacing joeschmo with your friend's actual Skype name. Save your entries and reload the dialplan when prompted.
SIP/joeschmo@127.0.0.1:5070

Security Warning. One final note of caution. Do NOT expose UDP port 5070 to the Internet unless you first secure this port with a username and password to avoid Internet intruders using your gateway as a free Skype dialing platform! You do not need 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your Asterisk server so we recommend you keep it securely behind at least a hardware-based firewall.

FreePBX Design. For those not using Incredible PBX, here is the FreePBX setup that Incredible PBX uses and that we recommend. For outbound Skype calls, you have two choices.

1. To place a call to a regular phone number using SkypeOut (which costs you money), you'll simply dial 8 plus the area code and number. Our foreign friends will have to adjust their dialplans and /siptosis/SkypeOutDialingRules.props accordingly. Today's setup assumes 10-digit phone numbers!

2. To place a call to a Skype username using a softphone that supports SIP URI dialing such as X-Lite, you simply precede the Skype username with an asterisk, e.g. *echo123 will connect you to the Skype Call Testing Service or *nerdvittles will connect you to the Nerd Vittles Skype demo.

For incoming Skype calls, the default setup routes those calls to a SIP URI: mothership@127.0.0.1. Whether you point this URI to an extension, ring group, or IVR is up to you. In the default Incredible PBX build, the mothership URI is pointed to the Stealth AutoAttendant, an IVR that plays a welcoming message and then transfers the call to a ring group if no digit is pressed by the caller.

Configuring FreePBX. To put this setup in place, use a web browser to access FreePBX on your Asterisk server. You'll need to create a Custom Trunk and then an Outbound Route.

1. Choose Setup, Add Trunk, Add Custom Trunk. Fill in the form so that it looks like the following using your own CallerID number obviously:

When you're finished, click the Submit Changes button and then reload the dialplan when prompted.

2. Next choose Setup, Outbound Routes, Add Route. Fill in the form so that it looks like this:

When you're finished, click the Submit Changes button. Be sure to move this new OutSkype route to the top position in your Outbound Routes listing in the right margin! Then reload the dialplan when prompted.

3. If you're not using Incredible PBX, add a new DayNight Control 1 option while you're still in FreePBX. Just specify where you want calls routed for Day mode and Night mode. Then, here's the easy way to activate SIP URI support on your Asterisk/FreePBX server. Copy the [from-sip-external] context from the extensions.conf file in /etc/asterisk. Now copy the content into extensions_override_freepbx.conf. Be sure to preserve the context name in brackets! On a FreePBX 2.8 system, make it look like the following. The additions we're making are shown in bold below:

[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => mothership,1,Goto(app-daynight,1,1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)

Finally, reload your Asterisk dialplan, and we're finished with Asterisk and FreePBX setup:

asterisk -rx "dialplan reload"

Fedora Builds. For those using recent Fedora builds, these systems have a full implementation of X-Windows and KDE. Just start the system in mode5 (graphics mode), log in, run Skype in one window and start up SipToSis in a terminal window using the commands in Step 7 above. Authorize external use of Skype when prompted.

Where To Go From Here. Well, those are the basics. You now can make one outbound Skype call at a time from your Asterisk server, and you can receive an inbound Skype call on any Asterisk extension when Skype users call your regular Skype name. If you want multiple Skype account support, then you'll need to do some tweaking. What you'll need is the STS Trunk Builder toolkit which is free, but proprietary. Enjoy!

Originally published: Tuesday, November 1, 2011

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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.

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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. Excerpt from the Skype Terms of Service: "Subscriptions are for individual use only. Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means). Each subscription is to be used for your own personal communication purposes only, to make calls to another individual. The use of the subscription for commercial gain, such as calling numbers specifically to generate income for yourself or others by placing such calls, is not permitted. Unusual call patterns may be considered indicative of such use and may result in us terminating your subscription and blocking your User Account in accordance with paragraph 11.2." [↩]

3 Steps to VoIP Nirvana: It’s Incredible PBX 2.0

Tuesday, August 30, 2011 / 8 Comments on 3 Steps to VoIP Nirvana: It’s Incredible PBX 2.0

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

We're pleased to introduce the latest and greatest Incredible PBX with an incomparable VoIP feature set. For the Pioneers, you now get transparent support for both Asterisk® 1.8 with PIAF-Purple and Asterisk 10 with PIAF-Red. Just download the PIAF 1.7.5.6.3 ISO and pick your favorite flavor. And this new release supports both FreePBX® 2.8 and 2.9. So explore away! Free Google Voice calling in the U.S. and Canada has been completely reworked to provide a simple FreePBX GUI menu that lets you set up one or many Google Voice connections in less than 10 seconds per line. We've also added Andrew Nagy's terrific EndPoint Manager that lets you configure dozens of SIP phones with the click of a button. You'll also find Kennonsoft's terrific new PBX in a Flash UI with HTML5 and CSS3 support for the latest Firefox, Chrome, and IE browsers. And now faxing with HylaFax & AvantFax works flawlessly using a dedicated (and free) Google Voice line. It doesn't stop there obviously. You still get almost every Asterisk application on the planet preconfigured and ready to use.

WARNING: Do NOT use Incredible PBX 1.8 or 2.0 with CentOS 6! There's a new Incredible PBX 3.0 with Incredible Fax 2.0 designed specifically for PBX in a Flash 2 with CentOS 6.2.

The Incredible PBX 2.0 Inventory. For those that have never heard of The Incredible PBX, here's the current 2.0 feature set in addition to the base install of PBX in a Flash with the latest CentOS 5.x, Asterisk 1.8 or 10, FreePBX 2.8 or 2.9, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Faxing, Hamachi VPN, and Mondo Backups are still just one command away and may be installed using the scripts included with base Incredible PBX 2.0 installation.

What began as a kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google's free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet and provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is still free, you'll never know you weren't connected via Ma Bell's overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play.

Just download the latest PBX in a Flash 1.7.5.6.3 ISO, burn to then boot from the PIAF CD, choose the PIAF-Purple or PIAF-Red Edition to load Asterisk 1.8 or 10 with FreePBX 2.8, and then run the new Incredible PBX 2.0 installer. In less than an hour, you'll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep you busy exploring for months.

Thanks to its Zero Internet Footprint™ design, Incredible PBX 2.0 remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.

Did we mention that all of this telephone goodness is still absolutely FREE!

Prerequisites. Here's what we recommend to get started properly:

Installing Incredible PBX 2.0. The installation process is simple and straight-forward. We're down to 3 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PIAF-Purple or PIAF-Red
2. Download & run Incredible PBX 2.0 installer
3. Configure Google Voice and a softphone or SIP telephone

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. To use Incredible PBX 2.0, just install the latest 32-bit version of PBX in a Flash. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.6 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the PIAF-Purple or PIAF-Red Payload to get the latest Asterisk 1.8 or Asterisk 10 release and all of the Google Voice goodies!

You can download the 32-bit PIAF from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin. See footnote 2 if you'd prefer to use a bootable thumb drive.

WARNING: Unless you boot from a flash drive, this install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve AND remove any USB flash drives! Press Ctrl-C to cancel.

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose PIAF-Purple or PIAF-Red option. Have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time. Log into your server with your root password and write down the server's IP address. You'll need it to access FreePBX with your browser.

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs, update-fixes, and passwd-master for you. So your system is relatively secure out of the box! See the Proxmox cautionary alert in the footnotes to this article!

Configuring Google Voice. You'll need a dedicated Google Voice account to support Incredible PBX 2.0. If you plan to use the inbound fax capabilities of Incredible PBX 2.0, then you'll want an additional Google Voice line that can be routed to the FAX miscellaneous destination using FreePBX. 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, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with The Incredible PBX. Google Voice no longer is by invitation only so, 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. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...

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

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

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

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

Incredible PBX 2.0 Installation. Log into your server as root and issue the following commands to download and run The Incredible PBX installer:

cd /root
wget http://incrediblepbx.com/incrediblepbx2.x
chmod +x incrediblepbx2.x
./incrediblepbx2.x

When The Incredible PBX install begins, you'll be prompted for your FreePBX maint password. This is required to properly configure CallerID Superfecta for you. Your credentials never leave your server!

Now have another 15-minute cup of coffee, and consider a modest donation to Nerd Vittles... for all of our hard work. 😉 You'll find a link at the top of the page. While you're waiting just make sure that you've heeded our advice and installed your server behind a hardware-based firewall. No ports need to be opened on your firewall to support Incredible PBX. Leave it that way!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

Logging in to FreePBX. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Users tab. It will change to Admin. Now click the FreePBX button. When prompted for a username, it's maint. When prompted for the password, it's whatever you set up as your maint password when you installed Incredible PBX. If you forget it, you can always reset it by logging into your server as root and running passwd-master.

Configuring Google Voice Trunks in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize your Incredible PBX to meet your specific needs. If you plan to use Google Voice, here's how to quickly configure one or more Google Voice trunks within FreePBX. After logging into FreePBX with your browser, 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. You can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill.

While you're still in FreePBX, choose Setup, Extensions, and click on the 701 extension. Write down your extension password which you'll need to configure a phone in a minute.

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.

Incredible Fax Installation. If you want the added convenience of having your Incredible PBX double as a free fax machine, run /root/incrediblefax.sh shell script when the Incredible PBX install completes. Plug in your email address for delivery of incoming faxes and enter your home area code when prompted. For every other prompt, just press the Enter key. For complete documentation, see this Nerd Vittles article. Don't forget to REBOOT YOUR SERVER when the install is finished, or faxing won't work!

Also be sure to set up a second, dedicated Google Voice number if you want support for inbound faxing. Once the Google Voice credentials are configured in FreePBX for the additional Google Voice line, simply add an Inbound Route for this DID to point to the FAX misc. destination that comes preconfigured with Incredible PBX 2.0. Just substitute your 10-digit Google Voice number for the DID number shown below. Save your entries and reload FreePBX.

Extension Password Discovery. If you're too lazy to look up your extension 701 password using the FreePBX GUI, you can log into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone or color videophone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

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 above. 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 your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

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 with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password. We're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX and uncheck the Disable Trunk checkbox. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.

Update: We've also included Trunk configurations for a dozen of our favorite hosting providers to get you started. You can sign up for service with any of them, insert your credentials in the existing trunk, uncheck the Disable Trunk checkbox, and then adjust your outbound route and add an inbound route for your new DID (if you get one).

Using ENUMPlus. Another terrific money-saving tool is ENUM. Your system comes with ENUMPlus installed. The advantage of ENUM is that numbers registered with any of the ENUM services such as e164.org can be called via SIP for free. You can read all about it in this Nerd Vittles' article. To activate ENUMPlus, you'll need to register and obtain an API Key at enumplus.org. It's free! Sign up, log in, and click on the Account tab to get your API key. Once you have your key, copy it to your clipboard and open FreePBX with your browser. Then choose SetUp, ENUMPlus and paste in your API Key. Save your entry, and you're all set. After entering your key, all outbound calls will be checked for a free ENUM calling path first before using other outbound trunks.

Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it's merely a greeting. To those "in the know," it's actually an AutoAttendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It's quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It's your phone bill. 😉

Configuring Email. You're going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Setup, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Setup, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. We also recommend you follow @NerdUno on Twitter. We'll keep you entertained and provide immediate notification of security problems that we hear about. Be safe!

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)

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PIAF Support Conference Bridge (Conf#: 1061)
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)

PBX in a Flash SQLite Registry. Last, but not least, we want to introduce you to the new PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here's the complete list of options:

  • A - Write the contents of the registry to registry.txt
  • B - PBX in a Flash install details
  • C - Extra programs install details
  • D - Update-fixes status and details
  • E - RPM install details
  • F - FreePBX modules install details
  • G - Incredible PBX install details
  • Q - Quit this program

And here's a sample from an install we recently completed.

Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. Please bear with us. To Mark Spencer, Malcolm Davenport, and the rest of the Asterisk development team, thanks for a much improved Asterisk. To Philippe Sultan and his co-developers, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer & Co., thanks for FreePBX 2.8 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made CentOS 5.6 and Asterisk 1.8 and 10 sit up and bark. No one will quite understand what an endeavor that is until they try it themselves. You've made it look so easy. And, finally, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.

Additional Goodies. Be sure to log into your server as root and look through the scripts added in the /root and /root/nv folders. You'll find all sorts of goodies to keep you busy. There's an all-new incrediblefax.sh script that painlessly installs and configures HylaFax and AvantFax for state-of-the-art faxing. The 32-bit install-cepstral script does just what it says. With Allison's Cepstral voice, you'll have the best TTS implementation for Asterisk available. ipscan is a little shell script that will tell you every working IP device on your LAN. trunks.sh tells you all of the Asterisk trunks configured on your system. purgeCIDcache.sh will clean out the CallerID cache in the Asterisk database. convert2gsm.sh shows you how to convert a .wav file to .gsm. munin.pbx will install Munin on your system while awstats.pbx installs AWstats. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. All the other scripts and apps in /root/nv already have been installed for you so don't install them again.

If you've heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud as well. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It's perfect for off-site backups and is included as one of the backup options in the PBX in a Flash backup utilities.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

Originally published: Wednesday, August 24, 2011

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Safely Interconnecting Asterisk Servers for Free Calling

Adding Skype to The Incredible PBX

Adding Incredible Fax to The Incredible PBX

Adding Incredible Backup... and Restore to The Incredible PBX

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with Information, Patches and Bug Fixes for Incredible PBX 2.0. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won't have to wait long for an answer to your question.


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. For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. [↩]
  2. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. If you are satisfied with PIAF-Purple running Asterisk 1.8, you also can install PBX in a Flash 1.7.5.6.2 from a USB flash drive. Just follow this Nerd Vittles article to get your thumb drive properly configured. Then it's plug-and-play. You also can run Incredible PBX 2.0 on a virtual machine such as the free Proxmox server. A security vulnerability has been reported in the Proxmox browser so be sure to run your server behind a secure, hardware-based firewall with no port exposure to the actual Proxmox server from the Internet. [↩]

PIAF 101: Taking Asterisk 10 for a Spin

Monday, August 29, 2011 / 8 Comments on PIAF 101: Taking Asterisk 10 for a Spin

There’s been some interest in a quick-and-dirty guide to get PBX in a Flash up and running without much in the way of bells and whistles. So here you go. This step-by-step will get PIAF-Red with Asterisk® 10 or PIAF-Purple with Asterisk 1.8.5.0 or 1.8.6.01 humming away. If you’re going to do things this way, then make sure your PIAF server or virtual host with PIAF is sitting behind a secure, hardware-based firewall (such as dLink’s Gaming Router) with NO INTERNET PORT EXPOSURE to your PIAF box!

UPDATE: Digium has dropped support for Google Voice in Asterisk 10 so we no longer recommend Asterisk 10 for production use. You can read all about it here.

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! And they can easily support a small 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 1.7.5.6.3 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 1.7.5.6.3 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. You’ll find OpenVZ and VMware templates on our download mirrors as well.

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 5.6 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. Just for fun, choose PIAF-Red which loads the latest Asterisk 10 beta. It works just fine!

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 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 PIAF 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 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.

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 last week’s 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, then… enjoy!

Originally published: Monday, August 29, 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.

 

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. See this Nerd Vittles article for Asterisk 1.8.6.0 install instructions. [↩]

Welcome to Frontier Days

Tuesday, July 26, 2011 / 2 Comments on Welcome to Frontier Days

One of my favorite vacations as a kid was spent enjoying Frontier Days in Cheyenne, Wyoming. If you’ve never been with your family, you need to add this to your Bucket List. It’s a week-long celebration that you’ll never forget. To commemorate this year’s event which is going on right now, we decided to celebrate by staging our own Frontier Days here at Nerd Vittles. It provides you an opportunity to join with us in kicking the tires of all the new stuff we’re working on this summer to write about in the fall. In the grand tradition of Cheyenne’s Frontier Days, expect a wild ride! If you’re a bit squeamish about knowing how sausage is made, today’s introduction to new projects may not be your cup of tea. For the pioneers, it’s Party Time! So let’s get started.

Introducing Asterisk 10. At the top of our list is the brand new Asterisk®, formerly known as Asterisk 1.10. You’ll want to read Kevin Fleming’s announcement of the name change, and then read Malcolm Davenport’s summarization of the new product. Here are a few excerpts:

A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters…

Asterisk 10 [also] provides basic video conferencing support. That’s right, if you and your friends have video-capable SIP devices, that all speak the same video codec and profile, you can create multi-party video conferences.

Asterisk 10 can also improve your faxing experience. Asterisk 1.4 is capable of T.38 pass-through, where one T.38 capable endpoint can send a fax directly to another T.38 capable endpoint – usually a couple of SIP peers. Asterisk 1.6.X and 1.8 are capable of T.38 termination, where Asterisk can read/write TIFF files from/to T.38 endpoints. Now, with Asterisk 10, transparency between non-T.38 and T.38 is possible.

Whenever there are major plumbing changes, there usually are some major surprises awaiting those of us that depend upon Asterisk to actually make calls. That’s where you come in. Tom King has quickly put together a new PBX in a Flash 1.7.5.6.3 ISO that includes PIAF-Red, aka the new Asterisk 10. We encourage you to try it on a non-production machine, and report any problems both to us (on the PIAF Forum) and to Digium® (in the Bug Tracker). Here’s a download link to get you started. Here’s the new Cepstral TTS installer.

Introducing Incredible PBX 2.0. Frontier Days wouldn’t be complete without a new version of Incredible PBX. In this beta release, we’ve reworked Google Voice support and added one of the most requested features, the ability to enter dial strings for trunks in outbound routes the old-fashioned way.

On the Google Voice front, we’ve replaced the hard-coded Google Voice code in Incredible PBX 1.8 with Marcus Brown’s new FreePBX® module. It not only makes Google Voice usage optional, but it also lets you add and remove multiple Google Voice trunks to your heart’s content. And the setup process takes less than a minute to enter your credentials.

Incredible PBX 2.0 also includes Andrew Nagy’s new Swiss Army Knife Module for FreePBX. This module adds some of the most requested features that currently are missing from FreePBX 2.8 and 2.9:

  • Export a CSV file of your Dial Patterns from Outbound Dial Plans
  • Use Textbox Dial Patterns for Outbound Routes
  • Modified Blacklist Module allowing any value, not just numbers
  • Coming Soon: reg-exp black/white list module

If you’d like to take Incredible PBX 2.0 for a spin, here’s a download link with instructions. Be aware that this version is NOT suitable for use on any system that is not also protected by a hardware-based firewall. For example, don’t use it on a hosted server such as RentPBX.com just yet. We use a different security model on hosted and cloud-based systems, and it is NOT included in this build. Finally, Incredible PBX 2.0 is not yet compatible with Asterisk 10 and PIAF-Red, but we’re working on it.

Introducing Google+. Unless you’ve been sleeping under a rock, you probably have heard that Google has a new little product of its own. In less than 3 weeks, Google+ has grown to over 20 million users, and it’s still by invitation only. You can read our writeup of it on Nerd Vittles. Suffice it to say, it is a game changer for those of us in the technology business. It’s an almost perfect tool for carrying on a problem-solving dialog, and we plan to make extensive use of it in coming months to support PBX in a Flash and Incredible PBX. Don’t be shy. We’ve got plenty of invites. All you have to do is drop us a note and include the word Google+ so we’ll know what you need. We’re turning requests around in less than a day. One final hint. Use your real name on Google Voice, or the Soup Nazi may remove your account. It’s become a bit of a brouhaha at the moment… as one might expect during Frontier Days.

Introducing OS X Lion. Apple has not been asleep at the wheel either. Their new operating system release is extraordinarily good but only available as an over-the-air update to an existing OS X 10.6.8 system. You can read our writeup of the gotchas for a quick and painless install. And, if you’re in the market for a new notebook, we can’t say enough good things about the new MacBook Air. It’s in a league of its own.

Introducing Google Chromebooks. Last but not least, we need to say a few words about the amazing new Chromebooks running Google’s Chrome OS. As with cellphones, Google is not making the hardware. So you have a choice of Samsung or Acer at the moment. The Samsung model starts at $429 for the WiFi only model. The comparable Acer machine is $80 cheaper. We opted for the Samsung WiFi machine which is well made, has an incredible battery life, and just works. For 95% of what we do, it’s a perfect device. There’s a short list of gotcha’s. First, you’ve got to have network connectivity since everything is cloud-based. Second, if your requirements include a lot of graphics manipulation and editing, this probably is not the machine for you quite yet. Finally, if movies (NetFlix) and music (Spotify) are must-have’s, you’d better wait a month or two until those products are available for the Chromebook. Google Music, which allows you to put your own music collection in the cloud, works fine today! There’s an add-on extension to Chrome for Google Voice. As of yesterday, it works flawlessly to make and receive calls. In summary, if your computing requirements primarily involve surfing the web, email, and SSH, then you’re going to be very happy with the Chromebook.

In our case, we’re trying to alternate our use between a Chromebook and the new MacBook Air. So far, we’ve been very satisfied with both. And the Chromebook is 1/4 the cost! Pioneers Forever! Enjoy!

Originally published: Tuesday, July 26, 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.

 

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…

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