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For those of you that missed the Dell SC440 at $199 last month and want to use the Nerd Vittles Orgasmatron II build for Asterisk®, now’s your chance. The machine is once again on sale for $299 until November 19. And now it includes a second hard disk. Still a steal. Be sure to upgrade to 2GB of RAM for the additional $19. How fitting for PBX in a Flash Birthday Week. Enjoy!

Some Recent Nerd Vittles Articles of Interest…

We began our 15-minute adventure last week with a turnkey install of Asterisk® onto a $199 Everex gPC2 using a fully-customized version of PBX in a Flash. If you haven't yet read the first article, start there. Today we want to cover what components are included and walk you through using most of them. When we're finished, you'll have a good idea why PBX in a Flash is not only different but also a quantum leap forward in the turnkey IP telephony marketplace. We'll also cover adding RAID 1 redundant drive support to your new server for about $40.

Putting Your Backup System Into Operation. Hopefully, you heeded our recommendation and purchased a $20 4GB USB flash drive to store backups of your new PBX in a Flash system. Cheap insurance! Now let's put it into production. In the /root folder of your new system, you'll find a PDF with complete documentation for the new Mondo Rescue backup system. If you flip to Appendix A, it will walk you through formatting your new flash drive for use with the backup software. Basically, you're going to delete the existing partitions on the drive, repartition it as a FAT32 partition (don't use the existing partition even if it says FAT32!!), and then reformat the drive. It takes under a minute to do it all. Once you're finished, let's initiate a backup just to be sure everything is working. Log into your server as root and type /etc/cron.weekly/disk-backup.cron. When the command prompt returns in about 30 minutes, type /root/usbcheck.sh to get a listing of the files on your USB flash drive. Now you can sit back and relax knowing that every Sunday night a new full system backup will be loaded onto your flash drive. Should something go horribly wrong with your main drive down the road, it's a simple matter to burn CDs of the ISO backup and reload everything, the same process you used to build your new system in the first place. Remember, we provided you a Mondo Rescue backup to build your system from ours so you know it works. For us at least, having automatic backups of your data is a critical component in any computer system, particularly your entire telephone system. While Asterisk aggregations are a dime a dozen these days, no one else has implemented any backup solution except PBX in a Flash.

Text-to-Speech on Steroids. The next thing you need to do is install Cepstral with Allison on your system. This gives something close to perfect text-to-speech capability for your entire phone system for under $25. And, yes, you can try it out first without spending a dime. Log into your server as root and type install-cepstral. Accept the defaults except create the missing directory when prompted. You're done. That was hard wasn't it. We'll test it out in a few minutes.

PiaF Software Update Service. At least for now, the PBX in a Flash Software Update Service continues to be a free option on all PBX in a Flash systems so by all means use it to keep your system current, bug-free, and secure. Log into your server as root and type update-scripts. Once the new scripts are loaded onto your system, type update-fixes. Yes, you can build an Asterisk system from many other ISO distributions. But you won't find another one that can keep your system current and secure without starting all over with a new ISO install. And when you want the latest and greatest version of Asterisk without missing a beat, that's easy, too. Just type update-source and have a cup of coffee while your system is upgraded. And don't forget to run update-fixes one more time to clean up any mess created by the upgrade.

Help at Your Fingertips. And, what if you forget all of these commands down the road and you're too lazy to pull out the documentation? Not to worry! Log into your server as root and type help-pbx.

What's Next? Now that you have a stable, secure, and up-to-date server, let's have some fun. We've loaded and preconfigured most of the Nerd Vittles applications in this build so all you have to do is learn the numbers to dial to use most of the applications. Here's a quick thumbnail sketch for each of the applications:

• AsteriDex RoboDialer and Telephone Directory
  This app gets you a phonebook, a web-based dialer using a browser or your cellphone, and a CallerID lookup source when used in conjunction with Ultimate CNAM. To add and update entries or lookup numbers, point your web browser to the IP address of your server: http://ipaddress/asteridex4/. For cell phone access, point the web browser on your cellphone to the public IP address or fully-qualified domain name of your server: http://publicIPaddress/cellphone/. Click on the link above for complete documentation and security suggestions.
• Telephone Reminders 4.0 with Support for Recurring Reminders and Web-based TTS Reminder Messages
  This app lets you schedule reminders for future events by telephone (dial 1-2-3) or with a web browser (http://ipaddress/reminders/). When the appointed date and time arrives, Asterisk swings into action and places a call to the number you designate to deliver a customized reminder message. Recurring reminders (daily, weekday, weekly, monthly, and annual) also are supported. And the text-to-speech web interface lets you schedule and deliver reminders using either Flite or Cepstral-generated messages with any web browser. For more info, click on the link above.
• NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS) - Dial 5-1-1
• Weather Reports by Airport Code (TTS) - Dial 6-1-1
• Weather Reports by ZIP Code (TTS) - Dial Z-I-P
• Worldwide Weather Forecasts (TTS) - Dial 6-1-2
• MailCall for Asterisk: Get Your Email By Telephone (TTS)
  This app reads your emails to you over the telephone. Some setup is required to plug in information about your email account. Once configured, dial 5-5-5 to retrieve your messages. Click on the link above for setup instructions.
• TeleYapper 4.0 Message Broadcasting System - Dial M-S-G (licensed for non-commercial use only!)
• CallWho for TTS Retrieval and Dialing of Entries in the AsteriDex Database (TTS)
  After entering contacts in AsteriDex, run http://serverIPaddress/asteridex4/dialcode.php to populate the dialcodes. Then dial 4-1-2 and enter the first three letters of anyone in your AsteriDex database to place a call.
• TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones - See setup instructions in last week's article

Of course, there are literally hundreds of things you can do with your PBX in addition to running the Nerd Vittles applications. Here's a short list of some of our favorites with some tips to get you started. The best source of information for more detail is our original article on PBX in a Flash 1.2 and the PBX in a Flash Forum.

• Stealth AutoAttendant with Welcome and Application IVRs
  Whenever an incoming call comes into your PBX, a generic greeting will play. If no button is pressed on the caller's phone, the call then will be routed to a ring group (700) for all of the extensions set up in that ring group. If no one answers, the call will be sent to the voicemail box for extension 701. While the greeting message is playing, the caller can press a digit on their phone to activate a hidden option in the Main IVR. As delivered, the only one that works is 0. This presents the caller with a list of Nerd Vittles apps from which to choose. You can add other options by modifying the Main IVR settings in FreePBX. To try out the Main IVR from any extension on your system, dial 7-7-7.
• Key Telephone Support Using Park and Parking Lot
  Most PBXs do not support shared line appearances like the old key telephones from Ma Bell. With these phones you could answer a call, place it on hold, and then someone else could pick up the call by pressing the blinking light on their phone. Our Aastra phone setup does much the same thing except, instead of placing a call on hold, you press the Park button. The parked extension number then will be read to you by Allison (starting with 71). Anyone else on your system can retrieve the parked call by pressing the ParkLot button on their Aastra phone and selecting the call to be retrieved by CallerID. Or, if the recipient knows the parking lot extension (e.g. 71), the recipient can pick up any phone and dial that extension number to retrieve the call.
• Intercom/Paging Support
  The Aastra phone setup for PBX in a Flash fully supports intercom calls and paging by pressing the ICom button on the phone. For more information, click Setup, Paging and Intercom from within the FreePBX web interface.
• Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
  Your system is preconfigured to support a USB Bluetooth dongle. No additional software installation is required. When properly configured, this lets you automatically forward your calls to your cellphone just by leaving your home or office with your Bluetooth-enabled cellphone. When you return, your calls will magically begin ringing on your local extension again. Click the link above for setup instructions.
• DISA
  Direct Inward System Access lets you call into your PBX and get dialtone to make an outbound call. To use it, you typically would add it as a hidden option on your IVR with a very secure password. We have preconfigured DISA support on your server. Just be sure you change the password to something very secure before activating it. To change the password, click Setup, DISA, DISAmain in FreePBX. Then save your changes and reload the Asterisk dialplan.
• Blacklisting with Web and Telephony Interfaces
  To block future calls from the last person who called you, dial *32. To block calls from a specific phone number, dial *30. To remove a number from the blacklist, dial *31. You also can use FreePBX to blacklist certain numbers. Just click Setup, Blacklist to access the web interface.
• CallerID Name Lookups from 8 Providers
  Most telephony providers reliably pass CallerID numbers but discard CallerID name info. With Ultimate CNAM which is preinstalled on your system, you can look up CallerID names from up to 8 different directory providers. To activate it, use FreePBX and click Setup, Inbound Routes, DefaultIncoming. Scroll down to CID Lookup Source and choose Ultimate CNAM from the dropdown box. Save your changes and reload the dialplan. For complete documentation, consult cnam_user_guide.pdf in the /root folder on your server. To choose the providers to use for the lookups, log into your server as root and type: cnam-config.pl
• Weekly Automated System Backups to a Flash Drive
  See the first section of today's article for the one-minute setup instructions.
• One Touch Day/Night Service
  With our Aastra phone setup, there is a DayNite button that toggles your system between Day and Night operation. As configured, the Night option transfers all calls to voicemail for extension 701. The Day option routes all incoming calls through the Main IVR which routes calls to the 700 Ring Group on timeout. To activate Night service from an Aastra phone, just press the DayNite button. To deactivate Night service, just press the button again. You also can dial *28 from any phone on your system to toggle Day/Night mode.
• Music on Hold
  Royalty-free music on hold is provided as part of the basic Asterisk install. Additional music can be added through the Music on Hold option in FreePBX. WAV files must be PCM Encoded, 16 Bits, at 8000Hz. See this thread for assistance. For other royalty-free and free music on hold, start here, choose Creative Commons for the License Type, and then click Go.
• Voicemail with Email Delivery of Messages and Pager Notification
  All of these settings are performed within FreePBX for each extension. Choose Setup, Extensions, and pick one of the extensions you already have created. Make certain that Voicemail Status is enabled. Then enter a valid email address and pager address. To include the voicemail as an attachment in the delivered email message, set Email Attachment to Yes. To include the CallerID in the voicemail message, set Play CID to Yes. To include the date and time of the call, set Play Envelope to Yes. To delete the voicemail message from the system after emailing it, set Delete Vmail to Yes. Don't ever do this until you're sure it's working reliably! If you want the option of calling back the caller when you retrieve your voicemail message by phone, set VMoptions to callback=from-internal. Submit your changes and reload the Asterisk dialplan to put the modifications into effect. If the emails are not delivered, then it may be because your ISP is blocking downstream SMTP traffic. To reconfigure your server to use gMail or Comcast as your SMTP host, click on one of the links.
• Voicemail Blasting
  This feature allows you to record a message and distribute it via voicemail to one or more extensions without actually calling the users. We've already configured extension 500 to send voicemail blasts to extensions 701 and 702. You can adjust the destinations in FreePBX by choosing Setup, Voicemail Blasts, Vmail (500). You also can add additional extensions to handle voicemail blasts to a different group of phones.
• Cell Phone Direct Dial
  There are two ways to make a cellphone an integral part of your PBX. The first involves setting up a specific extension for each cellphone and forwarding incoming calls to that extension to your cellphone. We've created extension 501 to show you how it's done. Once the extension is created, simply log into your server as root and issue a command like this where 6781234567 is your actual cellphone number:
  
asterisk -rx "database put CF 501 6781234567"

  When callers dial 501 on your system, your cellphone will automatically ring. Another option is to use FreePBX's Follow Me function under Setup. With this option, you can specify multiple destinations for incoming calls to a specific extension. Point to the Ring Strategy option and review the available choices. Choose the one that best meets your needs. Then enter the numbers to be called. Numbers outside your PBX should be in the format 6781234567# and must match your outbound dialing rules. You also can choose the time to attempt the call and what to do if no one answers. Very slick!
  
• Call Forward: All, Busy, No Answer
  While you can certainly use FreePBX's Follow Me functionality to accomplish any flavor of call forwarding, you also can dial codes from any extensions to activate call forwarding. To activate Call Forwarding All, dial *72; for Call Forwarding Busy, dial *90; for Call Forwarding No Answer, dial *52. To deactivate Call Forwarding All, dial *73; for Call Forwarding Busy, dial *91; for Call Forwarding No Answer, dial *53. To deactivate Call Forwarding All from a different extension, dial *74; for Call Forwarding Busy, dial *92. You also can activate and deactivate Call Forwarding from any Aastra phone using our default setup.
• Call Waiting
  To activate Call Waiting from any extension (which is the default), dial *70. To deactivate Call Waiting, dial *71.
• Call Pickup
  To pickup a call ringing on another extension, dial **.
• Zap Barge
  To barge into an existing call, dial 888.
• Call Transfer: Attended and Blind
  For attended call transfers where you can remain on the line until the other party answers, dial *2. For unattended call transfers, dial ## and then the number to which the call should be transferred.
• Dictation Service with Email Delivery
  Before using FreePBX's dictation service, you must activate Dictation Services for the specific extension to be used. Using FreePBX, go to Setup, Extensions, and click on the desired extension. Scroll down to the Dictation Services section of the form and enter your email address, the format of the sound files to be used, and change Dictation Service to Enabled. Save your settings and reload the dialplan. Then you can dictate your message by dialing *34. Once you finish your dictation, you can email it to your email address for this extension by dialing *35.
• Do Not Disturb
  To activate Do Not Disturb on any extension, dial *78. To deactivate Do Not Disturb, dial *79. There's also a button to accomplish the same thing with our Aastra phone setup.
• Phonebook Dial by Name - Dial 4-1-1
• VoiceMail Options
  To retrieve your voicemail from any phone, dial *97. To retrieve voicemail for a different extension, dial *98 or *98701 where 701 is the extension desired. To leave a voicemail message for any extension with voicemail enabled, dial *701 where 701 is the extension desired.
• Speed Dial
  To set up a user speed dial entry, dial *75. To call any previously established speed dial entry, dial *0 plus the speed dial number. To create or modify speed dial entries in FreePBX, click Tools, Asterisk Phonebook. You also can import entries from a CSV-formatted file.
• Flite and Cepstral Text to Speech (TTS)
  Flite TTS is installed by default with all PBX in a Flash systems using Asterisk 1.4. The Asterisk developers have broken Flite in the Asterisk 1.6-beta and refuse to fix it. Cepstral can be installed using the directions below with Asterisk 1.4 and 1.6-beta. To use Flite with Egor in your dialplan, here's the syntax:
  
exten => 444,5,Flite("Hello World.")

  To use Cepstral with Allison in your dialplan, use this syntax:
  
exten => 444,5,Swift("Hello World.")

• One-Click (almost) Cepstral TTS Install with Allison
  If our code didn't have a bug, you could have typed install-cepstral to install Cepstral. Instead, you need to type the following three commands:¹
  
sed -i 's|Cepstral_Allison-8kHz_x86-64|Cepstral_Allison-8kHz_i386|'↵
/usr/local/sbin/install-cepstral
sed -i 's|x86-64-linux|i386-linux|' /usr/local/sbin/install-cepstral
install-cepstral

  For detailed instructions on reconfiguring Nerd Vittles apps to use Cepstral instead of Flite, see this article. No software needs to be reinstalled. Simply change the dialplan and PHP app settings to use Cepstral as explained in the article. For more background on Cepstral, read this article. To register your newly installed Allison voice, go to this link. Be sure you select U.S. English language, Allison-8kHz voice, and Linux platform before you check out, or it's money down the drain. For 20% off your Cepstral license registration, use promo code “REALLUSIONTTS” when you check out. Write down the name, company (optional), and key that is issued once you fill in the blanks. Then log into your server as root, and type swift --reg-voice. Fill in the blanks with the information you wrote down above, and you're all set.
• Windows Networking with SAMBA
  Windows Networking with SAMBA is enabled by default in this special build. The default workgroup is "workgroup." To change the workgroup, log into your server as root and edit /etc/samba/smb.conf. Then restart SAMBA: service smb restart. You then can connect to your server from any computer that supports Windows networking using root as your username and whatever root password you created.
• Linux Firewall
  The IPtables firewall is enabled by default in all PBX in a Flash systems. For this build, we have enabled SAMBA access to your server. To disable it, log into your server as root, and issue the following commands:
  
cp /etc/sysconfig/iptables.nosamba /etc/sysconfig/iptables
service iptables restart

• WebMin
  WebMin is often described as the Swiss Army Knife of Linux. It provides a terrific web interface to Linux.everything. It is enabled by default in this install. To access it using a web browser, go to http://serverIPaddress:9001/ and login as root with the password you set up above for WebMin access. For complete documentation, go here.
• PBX in a Flash Software Update Service To Keep Your System Current
  To load current fixes for this build of PBX in a Flash, log into your server as root and type the following commands:
  
update-scripts
update-fixes

  To upgrade your system to the latest version of Asterisk 1.4, log into your server as root and type the following commands:
  
update-scripts
update-source
update-fixes


More Good News with the Everex gPC2. From the "Learn Something New Every Day Department," this news just in. The Everex gPC2 has built in hardware SATA RAID 1 support that actually works. What you'll need to get this going is a second 80GB hard disk to match the one delivered in your original box. Total cost: about $40. If one disk fails, the other kicks in automatically. Here's a link to purchase your drive. And here's the link that'll tell you how to get everything set up. Before you begin, make certain that you have a current ISO backup on your flash drive so that you can restore your system once the RAID setup is up and running. See the top of this article for the backup and testing procedure.

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. Join the text of a following line when you encounter this character: ↵ [↩]

Well, okay. We confess that today's creation doesn't quite measure up to the legendary Orgasmatron... but, look out Woody Allen, we're close. It's been a couple of years since we released our first preconfigured, turnkey Asterisk® install. Much has changed both in Asterisk and in the hardware and software environment since 2006. So today, to celebrate the six month anniversary of PBX in a Flash and the brand new PBX in a Flash 1.2 release, we're taking another stab at it. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... 15 minutes!

NOTE: This article and the Orgasmatron software have been updated. Click here to read the new article.

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

Our approach today is a little different than the last time around. The processing overhead of CentOS 5.1 has made VMware problematic. Luckily, the price of hardware has dropped like a rock. So today we're comfortable recommending the best phone, the best PC, and the best provider on the planet. And you'll still have your arms and legs intact after you pay the piper. If you've been following along with our articles, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i, and we've also identified a perfect small business/home computer on which to run a production Asterisk server for about 50 employees, the Everex gPC (aka "The WalMart Special"). Now that the second generation Everex gPC2 is readily available, we decided to preconfigure one of these systems from the ground up and then make a 2-disk ISO image backup of the whole system using Mondo. So, once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own Everex gPC2. But you must have a gPC2 for this to work so accept no substitutes, and don't try this with any other hardware or you'll end up with an Electronic Brick instead of an Orgasmatron. The $199 gPC2 systems are available from WalMart and NewEgg among others.

We've preconfigured outbound and incoming trunks from some terrific providers as well as some extensions on your new system. So you literally can sign up for service with these providers, plug in your phones, and you can be in full operation in under an hour. Our only word of caution is not to use these ISO images on another type of computer. Everything has been specifically tailored to the stock gPC2 and chances are very good that the install would not proceed much past erasing and reformatting your hard disk on a different flavor machine. We also recommend that, if you want to add our recommended $25 extra gig of RAM to the gPC2, hold off on installing it until after you have loaded the new ISO image. So... what do you get with this preconfigured build?

In addition to all of the goodness of a stock PBX in a Flash 1.2 build including Asterisk 1.4 running under CentOS 5.1 with all the latest and greatest versions of FreePBX, Apache, MySQL, and PHP, you also get 10 preconfigured Nerd Vittles applications for openers:

• AsteriDex RoboDialer and Telephone Directory
• Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
• NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
• Weather Reports by Airport Code (TTS)
• Weather Reports by ZIP Code (TTS)
• Worldwide Weather Forecasts (TTS)
• MailCall for Asterisk: Get Your Email By Telephone (TTS)
• TeleYapper 4.0 Message Broadcasting System
• CallWho for TTS Retrieval and Dialing of Entries in the AsteriDex Database (TTS)
• TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones

In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:

• Stealth AutoAttendant with Welcome and Application IVRs
• Key Telephone Support Using Park and Parking Lot
• Intercom/Paging Support
• Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
• DISA
• Blacklisting with Web and Telephony Interfaces
• CallerID Name Lookups from 8 Providers
• Weekly Automated System Backups to a Flash Drive
• One Touch Day/Night Service
• Music on Hold
• Voicemail with Email Delivery of Messages and Pager Notification
• Voicemail Blasting
• Cell Phone Direct Dial
• Call Forward: All, Busy, No Answer
• Call Waiting
• Call Pickup
• Zap Barge
• Call Transfer: Attended and Blind
• Dictation Service with Email Delivery
• Do Not Disturb
• Gabcast
• Phonebook Dial by Name
• Speed Dial
• Flite Text to Speech (TTS)
• Windows Networking with SAMBA
• Linux Firewall
• PBX in a Flash Software Update Service To Keep Your System Current
• One-Click Cepstral TTS Install with Allison... Just Type install-cepstral

Prerequisites. As mentioned, you'll need a $199 Everex gPC2 (WalMart or NewEgg ) to use this build. We also recommend an additional $25 gig of RAM for anything other than home use. We also recommend a 4GB USB flash drive on which to store automatic weekly backups of your new system. Finally, you'll need to cough up a whopping $5 to download the two-disk ISO image for this build. And, yes, we eat our own dog food. The ISO images you'll be downloading were captured as a backup on the flash drive of our gPC2 lab machine. If you use this special build, it seemed only fair that you cover the cost of the bandwidth to download it. As most of you know, we don't have the luxury of freeloading off SourceForge for our downloads. And we didn't want to impose upon our existing bandwidth providers to bring you this custom image. The good news is that, once you download the image from DreamHost, you are more than welcome to pass it along to one or more of your friends or business acquaintances at no charge. You can even do it electronically through the DreamHost Files Forever program. And, if you'd like to host this image for your fellow man at no cost, be our guest... and thank you! Bottom line: For about $250, you'll have the slickest, most reliable PBX on the planet with rock-solid weekly backups and, of course, the one-of-a-kind PBX in a Flash Software Update Service!

Getting Started. Once you have purchased your Everex gPC2, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. Having said that, we strongly recommend that you always keep your system running behind a NAT-based firewall/router. Almost any home router will do. Don't redirect any ports to the machine and don't turn the PC on just yet.

Download the two ISO images for the gPC2 from here. If you don't know how to create a CD from an ISO image, read that section from our article last week. In fact, read the whole article. It'll help you immensely down the road. Once you have the two CDs in hand, turn on the gPC2 and quickly insert Disk 1 into the CD/DVD drive and close the drive. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names, halt the install by pressing CTL-ALT-DEL and remove the CD. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 12 minutes. When prompted, insert Disk 2 and press the Enter key to finish the install. When the CD ejects, remove it and your gPC2 will reboot after you type exit.

After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear near the top of the screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot.

Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.

passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere


Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click Administration. Log in as admin:password. Then click Menu Config. Change Admin Pwd to a new password that you're NOT using elsewhere. Now click Update and then Done. Click Administration again and then Asterisk Mgmt (FreePBX). If you're prompted for username and password, use admin:password for now. After FreePBX loads, click Setup and then Administrators. In the far right column, click admin, fill in your new password, and click Submit Changes. Then do the same thing for maint. Finally, click on the orange Apply Configuration Changes button and then Continue with Reload. Whew!

Don't change any other passwords without first contacting us. Regardless of what you may read elsewhere, PBX in a Flash is now secure. If you want more details, read this article and this thread.

Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set.

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 (highly recommended). 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 (the size of a pack if cigs) known as a Sipura SPA-1001. It's under $60. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the SPA-1001 into your LAN, and then plug your phone instrument into the SPA-1001. Your router will hand out a private IP address for the SPA-1001 to talk on your network. You'll need the IP address of the SPA-1001 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The Sipura will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.

Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter 1234 as the Password, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Pick up the phone and dial 1234# to test it out.

Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with freePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.

Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.

Next, we need to tell your phone to use your new server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.

Log back into your server. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone and rename the file from 701.cfg to the MAC address of each phone.cfg. Do NOT use hyphens in the MAC address. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!

sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg


Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).

A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.

TCP 80 - HTTP (needed if you want to access the web sites on your new server from the Internet)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access, not recommended)
UDP 10000-20000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX communications typically between Asterisk servers)


Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world. Thanks to our friends at Vitelity, this is not only an easy process, but it's also an incredible deal... but only for PBX in a Flash users.

Vitelity: The Best Provider and Pricing on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up before the end of June, and you can purchase a Tier A DID with unlimited incoming calls for just $3.79 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price OR the call quality! Trust us. We've tried just about everybody. Update: This offer has been extended until July 15.

To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes.

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service.

An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a second outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.

First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.

If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.

Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.

Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:

canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=very
nat=yes
secret=yourpassword
type=peer
username=1092832198

For Incoming Settings, use from-pstn for the User Context and enter the following User Details:

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=very
nat=yes
type=user

For the registration string, enter a string like the following using your Peer Name and Password:

1092832198:yourpassword@did.voip.les.net/1092832198

Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Choosing a Preferred Provider. Finally, you'll need to decide whether to use AOL or Vitelity as your primary terminations provider. HINT: Vitelity is less costly. So we've set them up as your primary terminations provider with AOL as the backup. This is handled in FreePBX in the Outbound Routes tab under the AllCalls entry.

A Word About Mondo Rescue. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download this build and try it for yourself on your Everex gPC2. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.

Where To Go From Here. Well, we've covered a good bit of territory today so we're going to save the really fun stuff for our next installment. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy!

Continue reading Part II...

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.
 

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