Home » Posts tagged 'fax' (Page 11)

Tag Archives: fax

The Most Versatile VoIP Provider: FREE PORTING

Just 3 Steps to Paradise: It’s Incredible PBX for Asterisk 1.8

UPDATE: Incredible PBX 2.0 has just been released. Here's the article.

Hard to believe it's been over a year since we introduced The Incredible PBX. That makes today really special. And we're especially pleased to introduce a major facelift for the Incredible web site and, more importantly, an awesome new edition of Incredible PBX. Seems only fitting to release it on 5-9, a day synonymous with the level of perfection we're always shooting for. Time will tell. With the recent release of CentOS 5.6 came a new PBX in a Flash 1.7.5.6, and a much more stable Asterisk® 1.8.4.1.1 We've retweaked Incredible PBX to take advantage of the refinements and added some new features like faxing, SMS messaging, and MLB scores & schedules. Under the covers, you'll find Kennonsoft's incredible new PBX in a Flash UI with HTML5 and CSS3 support for the latest Firefox, Chrome, and IE8 browsers. Later this week, we expect one more iteration of the UI to conquer native Internet Explorer 9.2

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 ISO, burn to then boot from the PIAF CD, choose the Purple Edition to load Asterisk 1.8 and FreePBX 2.8, and then install the new Incredible PBX for Asterisk 1.8. In about 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 your head spinning for months.

Thanks to its Zero Internet Footprint™ design, The Incredible PBX 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 minimal 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!

The Incredible PBX Inventory. For those that have never heard of The Incredible PBX, here's a feature list of components you get in addition to the base install of PBX in a Flash the latest CentOS 5.x, Asterisk 1.8, FreePBX 2.8, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Fax, Hamachi VPN, and Mondo Backups are just one command away and may be installed using some of the PBX in a Flash-provided scripts.

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

Installing The Incredible PBX. 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 PBX in a Flash Purple Edition
2. Download & run The Incredible PBX 1.8 installer
3. Configure 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 for Asterisk 1.8, 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 new Purple Payload to get the latest Asterisk 1.8 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.

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 the install.

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 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. Write down the IP address of your new PIAF server. You'll need it to configure your hardware-based firewall in a minute.

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 secure out of the box!

Configuring Google Voice. You'll need a dedicated Google Voice account to support The Incredible PBX. 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 why take the chance. 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 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/incrediblepbx18.x
chmod +x incrediblepbx18.x
./incrediblepbx18.x

When The Incredible PBX install begins, you'll be prompted for the following:

Google Voice Account Name
Google Voice Password
Gmail Notification Address
FreePBX maint Password

The Google Voice Account Name is the Gmail address for your new dedicated account, e.g. joeschmo@gmail.com. Don't forget @gmail.com! The Google Voice Password is the password for this dedicated account. The Gmail Notification Address is the email address where you wish to receive alerts when incoming and outgoing Google Voice calls are placed using The Incredible PBX. And your FreePBX maint Password is the password you'll use to access FreePBX. It gets set automatically as part of the The Incredible PBX install. By the way, none of this confidential information ever leaves your machine... just in case you were wondering. 🙄

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 so leave it that way!

Here's a short video demonstration of the original Incredible PBX installer process. It still works just about the same way except there's no longer a second step to get things working.

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 last week's Nerd Vittles article. We should note that updated versions of HylaFax and AvantFax now have been incorporated into the installer thanks to gvtricks on the PIAF Forums, and Google Voice now seems to be much more reliable for delivery of faxes... if you happen to like FREE. 😉

Our experience suggests that using a single trunk for both voice and fax delivery is hit and miss so you may wish to consider adding an additional trunk just to support faxing. You'll find the templates for adding a second Google Voice trunk in the /tmp directory, and complete instructions are available on the PIAF Forums. We've also provided preconfigured trunk settings for both Vitelity and VoIP.ms if you'd like to try those options as well. Just plug in your credentials and configure an inbound route to map incoming faxes to the Fax Custom Destination. If you want to add support for a second Google Voice trunk, we've included dialplan2.txt and jabber2.conf in /tmp to get you started with the tutorial above.

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 Admin tab and choose FreePBX. 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.

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.

Adding Multiple Google Voice Trunks. Thanks to rentpbx on our forums, adding support for multiple Google Voice trunks is now a five-minute operation. Once you have your initial setup running smoothly, hop on over to the forums and check out this Incredible solution. You'll also find sample templates in the /tmp directory: dialplan2.txt and jabber2.conf.

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. 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.

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.



Click above. Enter your name and phone number. Press Connect to begin the call.


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 sit up and bark. No one will quite understand what an endeavor that was 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: Monday, May 9, 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 the latest 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. So you won't have to wait long for an answer to your questions.


Changes in PBX in a Flash Distribution. In light of the events outlined in our recent Nerd Vittles article and the issues with Asterisk 1.8.4, the PIAF Dev Team has made some changes in our distribution methodology. As many of you know, PBX in a Flash is the only distribution that compiles Asterisk from source code during the install. This has provided us enormous flexibility to distribute new releases with the latest Asterisk code. Unfortunately, Asterisk 1.8 is still a work in progress to put it charitably. We also feel some responsibility to insulate our users from show-stopping Asterisk releases. Going forward, the plan is to reserve the PIAF-Purple default install for the most stable version of Asterisk 1.8. As of June 1, Asterisk 1.8.4.1 is the new PIAF-Purple default install. Other versions of Asterisk 1.8 (newer and older) will be available through a new configuration utility which now is incorporated into the PIAF 1.7.5.6.2 ISO.

Here's how it works. Begin the install of a new PIAF system in the usual way by booting from your USB flash drive and pressing Enter to load the most current version of CentOS 5.6. When the CentOS install finishes, your system will reboot. Accept the license agreement, and choose the PIAF-Purple option to load the latest stable version of Asterisk 1.8. Or exit to the Linux CLI if you want a different version. Log into CentOS as root. Then issue a command like this: piafdl -p beta_1841 (loads Asterisk 1.8.4.1), piafdl -p 184 (loads Asterisk 1.8.4), piafdl -p 1833 (loads Asterisk 1.8.3.3), or piafdl -p 1832 (loads Asterisk 1.8.3.2). If there should ever be an outage on one of the PBX in a Flash mirrors, you can optionally choose a different mirror for the payload download by adding piafdl -c for the .com site, piafdl -d for the .org site, or piafdl -e for the .net site. Then add the payload switch, e.g. piafdl -c -p beta_1841.

Bottom Line: If you use the piafdl utility to choose a particular version of Asterisk 1.8, you are making a conscious decision to accept the consequences of your particular choice. We would have preferred implementation of a testing methodology at Digium® before distribution of new Asterisk releases; however, that doesn't appear to be in the cards. So, as new Asterisk 1.8 releases hit the street, they will be made available through the piafdl utility until such time as our PIAF Pioneers independently establish their reliability.



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. Unless you happen to own a Cisco 79XX phone. See comment below for details. []
  2. If you're using IE9, you'll need to run it in IE8 browser mode for the time being. We're working on it. 🙂 []
  3. For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. []
  4. 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. You also can run The Incredible PBX on a virtual machine such as the free Proxmox server. Another less costly (but untested) option might be this Shuttle from NewEgg: $185 with free shipping. Use Promo Code: EMCYTZT220 []

Incredible Fax: Free Faxing Returns to Incredible PBX 1.8

It’s been a rocky road getting an open source (free) faxing alternative to work reliably with Asterisk® 1.8. To further complicate things, CentOS 5.6 was finally released which brought us a few more Asterisk 1.8 headaches and updates finally leading up to an all-new and nearly perfect PBX in a Flash 1.7.5.6 thanks in large part to Tom King. The new release also forced some under-the-covers modifications in Incredible PBX. Now you’re caught up on last week’s news. But what have we done for you lately?

Well, one alternative was to shift gears to the commercial Fax for Asterisk from Digium® which is supported in FreePBX 2.8 and 2.9 and includes one free license. But we’re open source fans and, of course, nothing beats free. Thanks to the efforts of a number of folks on the PBX in a Flash forums including our old pal, Joe Roper, there is an alternative that folks have been wrestling with for over two years. The combination of Hylafax, Avantfax, and IAXmodem is a compelling open source solution if you don’t need T.38-compatible faxing.1 The drawback has been the learning curve to install all the components and get them working reliably together. Well, for those using Incredible PBX 1.8 with PIAF-Purple and Asterisk 1.8, today we have a newly minted installation script that is simple enough that even a monkey can use it. If you know your own email address and your local area code AND you can find the Enter key on your keyboard, you are fully qualified to perform today’s installation. It’ll take you under 5 minutes! We’ve also got a nice little surprise for you toward the end of this article.

Prerequisites. You’ll first need to install the latest version of PBX in a Flash with the PIAF-Purple (Asterisk 1.8) payload. Then sign up for a free Google Voice account and install Incredible PBX 1.8. You’ll find complete installation instructions for everything here. Can you just wing it and run this installation script on a garden-variety Asterisk 1.8 machine? No. And the reason is that all of these components have dependencies which are too complex to cover in a 5-minute article. You might want to have a look at the A-Fax Project which is where we started. Suffice it to say, the combination of PIAF-Purple and Incredible PBX 1.8 provides the ideal platform on which to install Incredible Fax. If you prefer to do-it-yourself, by all means have at it. We lost about 10 years worth of hair even starting with the work of a dozen very talented Linux gurus who have been wrestling with this for over two years! But, hey, YMMV! We never claimed to be the sharpest tool in the shed. 😉

Installing Incredible Fax. Once you have your Incredible PBX 1.8 platform up and running, adding Incredible Fax is a stroll in the park. Just log into your server as root and issue the following commands. If you’ve downloaded Incredible PBX in the last few days, the script may already be on your system. In this case, just type /root/incrediblefax.sh to run it.

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

After checking to make sure Incredible PBX 1.8 is installed, the script will prompt you to enter an email address where incoming faxes should be delivered. Then all of the necessary components will be installed after which the Avantfax install script will be run. With the exception of entering your local area code when prompted to do so, the correct response to every other question is to press the Enter key if you live in the U.S. or Canada. Don’t "improve" anything if you expect the end product to work reliably. For those outside North America, you’ll need to also make the usual adjustments to account for your country and city codes.

Avantfax has its own security model, but we’ve grown to appreciate the Apache authentication model which is built into PBX in a Flash so it’s been incorporated into Incredible Fax as well. When the install completes, just reboot your server to get everything working. On the PBX in a Flash web GUI, there will be a new Admin icon for Faxing. Or you can access Avantfax with a browser by going to http://serverIPaddress/avantfax. When prompted for your username and password, use maint and whatever your maint password happens to be. These can be reset with passwd-master. Literally everything has been preconfigured in Avantfax to get you going. Here’s a 3-minute video to show you how easy it is. Just don’t forget to reboot once the install completes.

If you want to be able to print to fax from Windows-based machines, then you’ll need to make one addition. Click on the small Toolbar icon in the upper right corner of the AvantFax home screen and choose New User from the pull-down Menu. For the user, enter Fax for the Name, fax for the Username, a secure password for Password, and an email address that is DIFFERENT from the one you used to set up Incredible Fax. Check the boxes for User Can Delete Faxes and User Can Fax From Any Modem. Finally, check the boxes for all four IAXmodems. Then click the Save button to add this new user.

A Word About Reliable Faxing. Suffice it to say that analog faxing over VoIP trunks is something less than ideal. If you want reliable analog faxing, then you’ll need a PSTN line from your favorite local telephone company. It doesn’t need any fancy add-ons like CallerID which doubles the price in many cities. Then you’ll need a properly configured analog telephone adapter (ATA) with at least one FXO port to support your Ma Bell phone line. Our favorite is the OBi110 which also can double as an additional Google Voice trunk for your PBX. But an SPA3102 will work equally well. It just costs more and gives you less.

Now that we’ve covered the obligatory warnings… will Incredible Fax work with a pure VoIP connection? Absolutely. We do it all the time. Is it flawless? No. Are there certain providers that are better than others? You bet. Do some providers not support faxing at all? Correct. Based on our 5+ years wrestling with this, here’s our recommendation. First, you’ll need a DID (i.e. phone number) from one of our recommended providers to handle inbound faxes. With the latest release of Asterisk 1.8, you no longer need a DID dedicated to faxing. In other words, you can use the same DID to receive incoming voice calls as well. The good news is that pay-as-you-go DIDs are dirt cheap. Some providers such as voip.ms offer DIDs for under $1 a month with 1¢ per minute calls. VoIP.ms also has unlimited inbound calling DIDs for under $4 a month. Other providers whose trunks we have found work reliably for VoIP faxing include Vitelity (see our special sign up deal below), Axvoice, Teliax, VoIPMyWay ($45 for first year with unlimited outbound and inbound calling with a local DID), and Future-Nine2. Google Voice trunks are hit and miss. We’re batting about .250 in our testing with Google Voice lines. Bottom Line: If VoIP faxing doesn’t work after you complete the install, it’s probably the fault of your VoIP trunk, not the setup. To make absolutely sure, connect a standard fax machine to an extension using an FXS telephone adapter and send a fax to that extension from the Avantfax web interface. You’ll find it works every time!

Configuring FreePBX for Incredible Fax. Here are the steps you’ll need to complete to get analog faxing working reliably with FreePBX. First, set up an account with one of the companies we’ve mentioned above. With voip.ms, create a subaccount on their site with credentials to use with the DID you purchased to link to that subaccount.

Unless you’re using today’s release of Incredible PBX, you’ll need to activate FreePBX’s Fax Configuration Module if you want to take advantage of Asterisk 1.8’s fax detection capabilities. It didn’t work reliably in previous Asterisk 1.8 releases. This module already is either available or already installed on your server. In the FreePBX GUI using a browser, choose Tools, Module Admin and then click on Fax Configuration. A drop-down list will provide several choices. Choose either Install or Enable depending upon the version of Incredible PBX you currently are running. Then click the Process button and finally Reload the settings when prompted.

Unless you installed Incredible PBX today, you’ll need to create a SIP trunk for your new provider in FreePBX using the credentials you set up on the provider’s web site. The VoIP.ms template now is included in Incredible PBX so you can just edit the existing one to add your credentials. And, at least with VoIP.ms, you can set the outbound CallerID to anything you like (as long as it’s legal). Unless you want a knock at your door, we wouldn’t recommend using the main number at the White House. Then put all of the settings below in the Outgoing Settings PEER Details where 1234567 is your main account number, subacctname is the name of the subaccount you created, and atlanta is your closest voip.ms server location:

username=1234567_subacctname
type=friend
trustrpid=yes
sendrpid=yes
secret=subacctpassword
nat=yes
insecure=port,invite
host=atlanta.voip.ms
fromuser=1234567_subacctname
disallow=all
context=from-trunk
canreinvite=nonat
allow=ulaw

For the registration string, it should look like the following. If you’re planning to only use the trunk for outbound faxing, then you can leave off the trailing DID number.

username:password@atlanta.voip.ms:5060/10-digit-DID

In addition to setting up the Trunk for your provider, you’ll also need to create an Outbound Route for sending faxes out through this trunk AND an Inbound Route to receive incoming faxes on the DID you purchased from your provider.

For the Outbound Route, we recommend setting the Dial Pattern with a prefix not otherwise used on your Incredible PBX so that you can make fax calls easily by dialing this prefix. For example, on our sample system, we used 7 so that fax calls could be made by dialing 7 plus a 10-digit number in the U.S. and Canada. Here’s how our Outbound Route for VoIP.ms looks in FreePBX, and the latest Incredible PBX release already has it in place as shown below:

For the Inbound Route, you want to specify the DID from your provider which must match the 10-digit number you affixed to the end of the trunk registration string above. If you don’t want to share this number for voice and fax calls, then simply direct these inbound fax calls to the Fax Custom Destination. Extension (329 spells F-A-X) also can be used to process incoming faxes and route them to your email address as well as the Avantfax web GUI.

Our experience suggests that using a single trunk for both voice and fax delivery is hit and miss so you may wish to consider adding an additional trunk just to support faxing. You’ll find the templates for adding a second Google Voice trunk in the /tmp directory, and complete instructions are available on the PIAF Forums. We’ve also provided preconfigured trunk settings for both Vitelity and VoIP.ms if you’d like to try those options as well. Just plug in your credentials and configure an inbound route to map incoming faxes to the Fax Custom Destination.

AvantFax in a Nutshell. Here’s a quick summary of the main features in the AvantFax web GUI. You can access the GUI by pointing a browser to the IP address of your server + /avantfax. After you enter your maint account name and maint password, the following screen will display with your Inbox. As noted, all of these incoming faxes also will be emailed to the account you set up when you ran the Incredible Fax install script.

The icons to the right of each thumbnail fax let you View, Rotate, Download PDF, Reply to Fax, Email PDF, Add a Note, Archive the Fax, and Permanently Delete the Fax.

At the top of the screen just to the right of Inbox is the option to Send a Fax. Here you’d specify the phone number to dial. Don’t forget the 7 and then a 10-digit number. Next you can attach a document from your local disk. Finally, fill in the blanks for the Fax Cover Sheet, and then click Send. Your fax will be on its way. You can monitor the progress of the fax transmission by clicking on Outbox. It’s also a good idea to fire up an SSH session to your server and run asterisk -rvvvvvvvvvv to monitor the first few calls to be sure all is well in Incredible FaxLand.

Where to Go Next. HylaFax and AvantFax are very mature open source products with a huge international following. We apologize for focusing primarily on U.S. and Canadian users today, but anything is possible with this software. The first piece you probably will want to tackle is adding Print to Fax capability on your Windows machine. The software you’ll need can be downloaded here. You’ll find excellent documentation on the setup by visiting the PBX in a Flash Forum. One little footnote for those using Windows 7. Microsoft and Apple are back to their old tricks so there are no Apple postscript print drivers in Windows 7. We’ve had equally good results using Dell’s 3100cn PS driver. Incidentally, there’s a similar print-to-fax utility for Mac OS X, but it’ll set you back $36. Here’s the link. HylaFax also maintains a terrific resource list for those that want additional goodies for PCs, Macs and Linux systems.

Originally published: Monday, May 2, 2011


Changes in PBX in a Flash Distribution. In light of the events outlined in our recent Nerd Vittles article and the issues with Asterisk 1.8.4, the PIAF Dev Team has made some changes in our distribution methodology. As many of you know, PBX in a Flash is the only distribution that compiles Asterisk from source code during the install. This has provided us enormous flexibility to distribute new releases with the latest Asterisk code. Unfortunately, Asterisk 1.8 is still a work in progress to put it charitably. We also feel some responsibility to insulate our users from show-stopping Asterisk releases. Going forward, the plan is to reserve the PIAF-Purple default install for the most stable version of Asterisk 1.8. As of June 1, Asterisk 1.8.4.1 is the new PIAF-Purple default install. Other versions of Asterisk 1.8 (newer and older) will be available through a new configuration utility which now is incorporated into the PIAF 1.7.5.6.2 ISO.

Here’s how it works. Begin the install of a new PIAF system in the usual way by booting from your USB flash drive and pressing Enter to load the most current version of CentOS 5.6. When the CentOS install finishes, your system will reboot. Accept the license agreement, and choose the PIAF-Purple option to load the latest stable version of Asterisk 1.8. Or exit to the Linux CLI if you want a different version. Log into CentOS as root. Then issue a command like this: piafdl -p beta_1841 (loads Asterisk 1.8.4.1), piafdl -p 184 (loads Asterisk 1.8.4), piafdl -p 1833 (loads Asterisk 1.8.3.3), or piafdl -p 1832 (loads Asterisk 1.8.3.2). If there should ever be an outage on one of the PBX in a Flash mirrors, you can optionally choose a different mirror for the payload download by adding piafdl -c for the .com site, piafdl -d for the .org site, or piafdl -e for the .net site. Then add the payload switch, e.g. piafdl -c -p beta_1841.

Bottom Line: If you use the piafdl utility to choose a particular version of Asterisk 1.8, you are making a conscious decision to accept the consequences of your particular choice. We would have preferred implementation of a testing methodology at Digium before distribution of new Asterisk releases; however, that doesn’t appear to be in the cards. So, as new Asterisk 1.8 releases hit the street, they will be made available through the piafdl utility until such time as our PIAF Pioneers independently establish their reliability.


Need help with Asterisk? Visit the PBX in a Flash Forum or Wiki.
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. Yes, we’re aware that HylaFax theoretically supports T.38 with the right hardware. Feel free to point us to someone who has it actually working with Asterisk 1.8. 🙄 []
  2. Vitelity, Teliax, VoIPMyWay, and Future-Nine trunks require the following additional entries in your Inbound trunk settings: t38pt_rtp=no, t38pt_tcp=no, t38pt_udptl=no []

Asterisk on Steroids: The Orgasmatron Installer, Part IV

If you haven't installed our two dozen turnkey Asterisk® applications in under 5 minutes, it's not too late! We recently introduced our Orgasmatron Installer for PBX in a Flash. And today we wrap up the tutorials with Part IV in this series. Faxing and email work out of the box. More than a dozen extensions and a number of hosting provider trunks are preconfigured. Delivery of CallerID names with numbers is available from over a dozen providers of your choice. ODBC database connectivity is now painless. And the Flite text-to-speech engine is preconfigured with Cepstral TTS only a few keystrokes away. Also included are FreePBX 2.5, Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Here's the complete list of what 5 minutes of your time brings to this one-of-a-kind Asterisk server platform:

In Part II of this series, we walked you through securing your system and configuring a few of the major applications: AsteriDex, CallerID Superfecta, CallWho, Cepstral, and Emailing with SendMail. In Part III, we covered faxing with nvFax, FONmail, FreePBX backups, the Gizmo5 FreePBX module, setup of Hamachi VPNs, interconnecting Asterisk servers with IAX, setting up on-the-fly conferences, ODBC database implementation, and telephone reminders using a phone or web browser. Today, we'll cover the remaining applications in the Orgasmatron build: Hotel-Style Wakeup Calls, Mondo Full System Backups, Yahoo Newsclips, SIP URI support, TeleYapper, Tide Reports with xTide, Weather Reports by telephone, and how to use the Zaptel updater.

Hotel-Style Wakeup Calls. This application was specifically designed for FreePBX and does just what the name implies. From any phone connected to your PBX, dial *68 and follow the prompts using 4-digit numbers for the desired wake up call times. Then wait for your wakeup call. Doesn’t get much easier than that. There are a number of configuration options which can be set by logging into FreePBX and choosing Admin, Tools, Wakeup Calls. Operator mode lets you specify extensions which can set up wakeup calls for any extension. You also can define the ring time, number of retries, and the time to wait between retries. For the complete tutorial, see this Nerd Vittles article.

Mondo Full System Backups. One of the age-old limitations of Asterisk@Home and now trixbox was the inability to make a full disk backup of your PBX so that it could be restored after a catastrophic event, man-made or otherwise. Tom King solved all of that with his implementation of Mondo Rescue for PBX in a Flash systems. There are numerous options for storing the backups. We prefer using a USB flash drive and rotating between two of them. With falling prices of flash drives, you now can purchase 8GB and 16GB models for peanuts. To enable the backup system, insert a USB flash drive on your PBX. Log into your server as root and type dmesg. Scan through the contents of the display until you find the device name for your USB flash drive. The listing should look something like this:

usb-storage: waiting for device to settle before scanning
Vendor: Kingston Model: DataTraveler 2.0 Rev: PMAP
Type: Direct-Access ANSI SCSI revision: 00
SCSI device sdc: 15874048 512-byte hdwr sectors (8128 MB)
sdc: Write Protect is off
sdc: Mode Sense: 23 00 00 00
sdc: assuming drive cache: write through
SCSI device sdc: 15874048 512-byte hdwr sectors (8128 MB)
sdc: Write Protect is off
sdc: Mode Sense: 23 00 00 00
sdc: assuming drive cache: write through
sdc: sdc1
sd 8:0:0:0: Attached scsi removable disk sdc
sd 8:0:0:0: Attached scsi generic sg2 type 0
usb-storage: device scan complete

In the listing above, it would tell you that your device is named sdc1. In Mondo parlance, this device name would be /dev/sdc1. Your mileage may vary obviously depending upon the type server you are using. Don't guess! Otherwise, you may end up inadvertently formatting (aka erasing) your primary hard disk since this is the first step in the Mondo backup process.

Once you are positive that you have the correct device name for your flash drive, edit /etc/asterisk/disk-backup.conf. Change line 11 to the following: CONFIGURED="1". Then change line 50 to the device name for your flash drive: USBDEVICENAME="/dev/sdc1". Save your changes. Now run a test backup to be sure everything is working properly: /etc/cron.weekly/disk-backup.cron.

You can review the contents of your flash drive by making a script with the following commands. Be sure to make the script executable and use the actual device name for your flash drive:

#!/bin/bash
mount -t vfat /dev/sdc1 /mnt/usbmondo
df /mnt/usbmondo
echo " "
ls -all -h /mnt/usbmondo
umount /mnt/usbmondo

Be aware that Mondo backups may not properly restore on some of the new Atom-based netbooks. A patch has been released by the Mondo development team which we currently are testing. This newer version also supports creation of bootable flash drives as part of the backup process. Stay tuned.

Yahoo Newsclips for Asterisk. This was one of the first Nerd Vittles text-to-speech (TTS) applications for Asterisk, and it remains one of the most popular. To use it, dial 511 from any phone on your Asterisk system. The default setup gives a choice of numerous Yahoo news and sports feeds which will be read to you over the telephone. For detailed setup instructions, see the original Nerd Vittles article. The application, by default, uses the Flite text-to-speech engine. If you have purchased Cepstral, you can easily reconfigure Newsclips for Asterisk to use Cepstral as the TTS engine. Just edit nv-news.php in /var/lib/asterisk/agi-bin and change the $ttspick entry in line 16 from 0 to 1.

Asterisk SIP URI Support. Direct SIP-to-SIP communications is one of the most exciting emerging trends in Internet telephony. Within 10 years, Gartner predicts that 50% of all phone traffic will be pure IP from end to end. You can start using it with your new server to make free phone calls today. All that's really needed is a SIP URI for your server. SIP URI's work just like email addresses except they tell phone systems where to deliver calls over the Internet. The Orgasmatron build preconfigures a number of SIP URI's for you including mothership, e164, and fax. This means that anyone can contact you by "dialing" your SIP URI using either the IP address of your server or a fully-qualified domain name that points to that IP address. A typical SIP URI would look like this: mothership@192.77.210.14. This tells the calling system to route the call to the mothership context on the Asterisk server living at 192.77.210.14. You also can contact the demo applications on your server by dialing nv-demo@192.77.210.24.

The next logical step with SIP URI's is to interconnect your server with a traditional POTS phone number using your SIP URI. You can sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) or IP address of your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and ipkall.com will beam you a Washington state phone number within a day or so. Just use it at least once a month, and you've got free inbound calls using a real telephone number forever. You can do much the same thing with Gizmo by signing up for an account using the FreePBX web interface included in the Orgasmatron build. You can't beat the price! For more detail on SIP proxies, see this Nerd Vittles article. To add your new number to directory assistance listings in the United States, just go to listyourself.net and sign up.

The other great use for SIP-to-SIP communications is to register yourself in the ENUM system so that other Asterisk and FreeSwitch systems can translate your plain old telephone number into a SIP URI and place the call SIP-to-SIP without any communications charges. To sign up for the service, go to both 164.org and enumplus.org. It only takes a minute. ENUM is implemented for default outbound calls by default on Orgasmatron builds. This means your server will attempt to place the call for free through ENUM before using your other outbound trunks for which you have to pay a fee to a provider.

TeleYapper. This application is an automated message broadcasting service commonly known as a call blasting or phone blasting system. It is licensed for non-commercial use including the following: to send prerecorded phone messages for neighborhood association announcements, school closings, tornado alerts, little league practices, fund raisers, municipal government reminders, and for just about any other non-commercial purpose. TeleYapper is simple to use. Dial extension M-S-G (674) on your Asterisk system and enter your password. You'll be prompted to record a message. Next you enter the group number for delivery of your TeleYapper message. The system will tell you how many recipients are in the group you have chosen. You then can begin the phone blasting session, or you can choose to resend messages to failed calls on a previous try to the same group. TeleYapper keeps track of which calls were successfully delivered and which were not so that follow-up calls can be placed. For detailed instructions on how to add entries to your TeleYapper database, see this Best of Nerd Vittles article.

Tide Reports with xTide. As the name implies, the xTide for Asterisk TTS application lets you retrieve tide and lunar information about any U.S. port by dialing 8433 (T-I-D-E) from any phone connected to your Asterisk system.

The default port setting for xTide for Asterisk is Pawleys Island, South Carolina. You can change this to meet your needs. There are three steps to reconfiguring the desired port city. First, identify a port city supported by xTide. Second, test the port city you have chosen using the tide application. Third, configure xTide for Asterisk for your desired port city. To identify whether a particular port city is supported by xTide, visit this link and search for the city you wish to use. Once you have verified that your desired location is supported, test it manually with the tide application that was installed as part of xTide for Asterisk. Log into your server as root and issue the command: tide -l "portcity", e.g. tide -l "boston".

Once you have verified that you get a tide report for your chosen city, simply reconfigure xTide for Asterisk to support that destination. While still logged in as root, edit /etc/asterisk/xtide.conf and change the contents to your new city. Be careful NOT to add any blank lines to the config file!

SITE="pawleys"
SITENAME="Pawlees Island, South Carolina"

You'll note that the spelling of the SITENAME was modified slightly to assist the TTS application. Complete details for configuring xTide for Asterisk as well as instructions for changing to Cepstral TTS support are included in the Best of Nerd Vittles article.

Weather Reports by Phone. Three separate TTS weather applications are included in the Orgasmatron build. You can retrieve weather forecasts by zip code and airport code as well as by international city. Dial Z-I-P and enter a 5-digit zip code. Or dial 6-1-1 and enter a three-character U.S. airport code. Or dial 6-1-2 and choose the international city preconfigured in your system. By default, the Worldwide Weather Forecasts for Asterisk application comes preconfigured to support 10 cities around the world. Here's the list:

0 - Tokyo
1 - Washington
2 - Berlin
3 - Florence
4 - Gough Island
5 - London
6 - Moscow
7 - Sydney
8 - Toronto
9 - Zurich

For details on changing the city codes as well as tips in using the other weather applications, see the Best of Nerd Vittles articles.

Miscellaneous Scripts. For your convenience, a script is included to update your zaptel setup whenever you add a card to your system or install a new Linux kernel. You'll find the script in the /root/nv folder on your server: zaptel-update.sh. There's also a script to install A2Billing: install-a2billing. There's also a detailed FAQ to walk you through configuring the Amazon S3 cloud computing service to work with PBX in a Flash as an off-site storage facility: s3cmd.faq. For configuration tips on configuring S3, see this Nerd Vittles article.

CallerID Superfecta 2.1. It's only been 10 days since the new FreePBX-based CallerID Superfecta was released. But wait until you see this new version. The original release of this application included 3 data sources. This one has 15 including the first Canadian source! There are too many new features to mention all of them here, but here's the short list:

1. Added Local Caching to MySQL
2. Retention of Valid Caller ID Name if Provided by Trunk
3. "Automatic" Support for sources requiring authentication
4. Post CID retrieval processing for source scripts
5. Altered whocalled behavior to return textual CallerID info
6. Support for sources with CID and SPAM rankings
7. Enhanced script error reporting in debug interface
8. "Report Back" capability to populate Data sources

You'll have to install this yourself unless you downloaded the Orgasmatron Installer (v1.4) after 5 pm EDT yesterday, May 24. The install instructions are included in the release notes, and it only takes a few seconds. Here's a link to the writeup on the new module on the PBX in a Flash Forum.

Unrelated But Still Interesting. If you're fascinated by all discoveries relating to words beginning with the letters o-r-g-a-s-m, be sure to check out Mary Roach's recent Ted Talk. Enjoy. 🙂

Read Part I and download the software.

Read Part II.

Read Part III.



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


Twitter Magic. If you haven't noticed the right margin of Nerd Vittles lately, we've added a new link to our Twitter feed. If you explore a little, you'll discover that the user interface now brings you instant access to every Twitter feed from the convenience of the Nerd Vittles desktop. Enjoy!


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...

Asterisk on Steroids: The Orgasmatron Installer, Part III

Happy Cinco de Mayo! And you can celebrate the event by installing two dozen turnkey Asterisk® applications in under 5 minutes! We recently introduced our new Orgasmatron Installer for PBX in a Flash. And today the saga continues with Part III in our series. Faxing and email work out of the box. More than a dozen extensions and a number of hosting provider trunks are preconfigured. Delivery of CallerID names with numbers is available from a half dozen providers of your choice. ODBC database connectivity is now painless. And the Flite text-to-speech engine is preconfigured with Cepstral TTS only a few keystrokes away. Also included are FreePBX 2.5, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Here's the complete list of what 5 minutes of your time brings to your Asterisk server platform:

In Part II of this series, we walked you through securing your system and configuring a few of the major applications: AsteriDex, CallerID Superfecta, CallWho, Cepstral, and Emailing with SendMail. Today, we'll tackle nine more applications in the list.

Fax Module with nvFax. The NVfax module provides basic incoming and outgoing fax functionality for your PBX in a Flash system. It's not perfect because faxing with VoIP providers is hit and miss at best! As installed, inbound faxing works after a simple configuration. Here are the three steps:

#1. Log into your server as root and edit fax-process.pl in the /var/lib/asterisk/bin folder. Change the following default parameter to make it your default MAILTO email address:

my $to = "JoeSchmoe\@gmail.com";

NOTE: Always edit system files like this: nano -w filename

#2. Using a web browser, log into FreePBX and choose Admin, Setup, General Settings. In the Fax Machine section of the form, choose system as the extension for receiving faxes, enter the destination email address for incoming faxes, and enter an email from address for outbound faxes.

#3. While still in FreePBX, you need to define how you want faxes processed when they are received from outside your PBX. Choose Admin, Setup, Inbound Routes. For each incoming route on your PBX where you want to enable receipt of faxes, click on that incoming route definition. In the Fax Handling section of the form, choose system as the fax extension, enter the fax email destination address, choose nvfax as the fax detection type, and use 5 as the fax detection delay setting. Save your settings for each inbound destination and then reload your dialplan.

You can test it by plugging a real fax machine into a VoIP phone adapter such as the Linksys SPA-2102 and assigning the ATA an extension number on your PBX. Using the fax machine, simply send a fax to extension 329 (F-A-X). It should arrive as a PDF in your email inbox within a couple minutes.

Once you get fax delivery of faxes from inside your PBX working reliably, then you're ready to graduate to the Big League and get faxing from outside your PBX working. This is 99% dependent upon the quality of inbound calls from your DID provider. If your DID provider doesn't support ULAW, give up or switch providers. We have successfully tested inbound faxing with TelaSIP, Teliax, voip.ms, and Future-Nine. With Teliax and Future-Nine, you will need to add the following settings to your Incoming Trunk Configuration in FreePBX:

t38pt_rtp=no
t38pt_tcp=no
t38pt_udptl=no

For additional tips and tricks, read our Best of Nerd Vittles article on faxing.

FONmail for Asterisk. FONmail is one of several applications that works in conjunction with AsteriDex. It lets you pick up a telephone connected to your Asterisk system, dial 6245 (M-A-I-L), and dictate a message for email delivery to someone in your AsteriDex database. You'll be prompted for the phone number of your recipient, or you can look up a person using the first three letters of their name in the AsteriDex database. Once you record your message and choose the recipient, the dictated message is emailed to the recipient using the email address you've entered for that person in AsteriDex.

For FONmail to work, you obviously have to add entries into AsteriDex (with email addresses) for the recipients you intend to select, and you need to populate the new dialcodes for AsteriDex by following the instructions in Part II of this tutorial. The final piece is specifying your return email address for the outbound emails. Set your return email address by editing the $email entry at the top of nv-mailit.php. The file is stored in /var/lib/asterisk/agi-bin.

FreePBX Backups. A disaster recovery plan is a critical component with any computer system, and PBX in a Flash is no different. You need to have a plan for recovering from a disaster whether that disaster is an Act of God, or man-made, or the result of a hardware failure. Our recommended strategy goes like this. Make weekly full disk backups with Mondo to at least a pair of USB flash drives. Replace the drive each week and take the other drive off site. In addition, make daily or weekly FreePBX backups and copy them to a safe place. Amazon S3 offers a convenient, inexpensive off-site storage facility for FreePBX backups. FreePBX backups let you restore FreePBX components to a machine state at the time the backup was made. Here's how to set up FreePBX automatic backups. Be sure you clean out old backups from time to time as they take up disk space. The backups are stored in folders under /var/lib/asterisk/backups based upon the name you assign to your backup schedule.

Here's how to set one up to make a backup on demand:

1. Open FreePBX with your web browser.
2. Choose Admin, Tools, Backup and Restore, Add Backup.
3. Give the backup schedule a name, e.g. RightNow.
4. Change all Radio buttons to Yes to backup everything.
5. Backup schedule: Run Backup Now.
6. Click Submit Changes button to kick off the backup.

Here's how to set one up to make a weekly backup every Sunday night:

1. Open FreePBX with your web browser.
2. Choose Admin, Tools, Backup and Restore, Add Backup.
3. Give the backup schedule a name, e.g. Daily.
4. Change all Radio buttons to Yes to backup everything.
5. Backup schedule: Run Backup Weekly (on Sunday).
6. Click Submit Changes to save new backup schedule.

Gizmo5 FreePBX Module. One of the VoIP providers that provides enormous flexibility in getting the most out of your new system is Gizmo5. For very little money and virtually no configuration hassles, Gizmo5 can't be beat. One of the slick functions that Gizmo5 provides is the ability to make 5-minute phone calls to any Skype user at no cost. For $20 a year, you can make as many 2-hour Skype calls as you like to your ten best friends. For more details, see our article. The Orgasmatron installer puts everything in place for you to set up a Gizmo account quickly from within the FreePBX interface. Just choose Admin, Setup, Gizmo5 Integration. Just follow the prompts to create your new account and make an initial deposit.

Installing the Hamachi VPN. Once you've run the Orgasmatron Installer, you have the option of installing the Hamachi virtual private network (VPN) which supports the interconnection of 16 computers at no cost. Simply run the install-hamachi.x script which you'll find in your /root/nv folder. For complete configuration instructions, read the install-hamachi.pdf file and hamachi.faq, both of which are also in the same directory.

Interconnecting Asterisk Servers with IAX. If you don't plan to interconnect your Asterisk server with one or more other Asterisk servers, then delete the Remote-Host outbound route in FreePBX and then delete the remote-peer trunk. If you plan to use the ODBC demo examples on extensions 222 and 223, you at least will need to change the Dial Pattern for the Remote-Host outbound route by deleting the 2XX entry as explained elsewhere in this article. What this provided was a simple way to interconnect extensions in the 200-299 range of numbers on a remote PBX.

If you do plan to interconnect Asterisk servers, then change this 2XX Dial Pattern to match the extension numbers on your remote PBX. For example, if the remote Asterisk server uses extensions in the 7000-7999 range of numbers, you'd want to include a 7XXX entry in your Remote-Host Dial Pattern.

To enable, interconnection of your new server to another Asterisk server, edit the remote-peer trunk and insert the actual IP address of your remote host. Also change the secret in the Peer and User sections to a very secure entry and use the same secret entry in your remote host trunk setup.

On the remote server, create a new IAX trunk with settings like the following using your correct secret and the IP address of your new server that was built with the Orgasmatron Installer:

MeetMe Conferences On the Fly. If you're accustomed to spending hundreds of dollars to schedule and run phone conferences with dozens of people, those days are officially over with PBX in a Flash. You now can purchase a phone number in 2600+ rate centers in the United States with support for 20 simultaneous calls for under $9 a month. Once you have purchased your DIDforSale DID and configured the new trunk on your server, simply point the inbound route for that trunk to Misc Destination: MeetMe CONF.

To set up a conference at any time, pick up any phone on your PBX and dial 2663 (C-O-N-F). When prompted for the conference number, make one up, e.g. 30303. When prompted for a conference PIN, make one up, e.g. 1234. Now notify all conference participants to dial the Conference DID (or 2663 for internal users) and to use 30303# for the conference number and 1234# for the PIN. When everyone hangs up, the conference ends. Simple as that!

ODBC Database Connectivity. All of the necessary components to support ODBC database integration with Asterisk have been installed for versions of the Orgasmatron Installer after May 1. Also included are two sample dialplan components that demonstrate how to build ODBC applications. These two samples are explained in the Nerd Vittles ODBC article. The extensions used by these two samples are 222 and 223. If you used an older version of the Orgasmatron Installer, you'll have to manually add ODBC support and the sample extensions conflict with the default routing rules for interconnecting your server to another Asterisk server. So you have two options. Either change the Dial Pattern for interconnecting to the remote server by deleting the 2XX entry or modify the extension numbers for the ODBC demos in /etc/asterisk/odbc.conf. Once you have addressed this inconsistency, you can activate the ODBC demo applications by inserting the following line in the [from-internal-custom] context of extensions_custom.conf in /etc/asterisk: #include odbc.conf

Then reload your Asterisk dialplan: asterisk -rx "dialplan reload"

Reminders by Phone and by Web. The latest version of the Best of Nerd Vittles Telephone Reminders 4.0 application is included in the Orgasmatron Installer. You can schedule reminders by telephone by dialing 1-2-3 from a phone connected to your Asterisk PBX. The default password is 12345678. To keep strangers from using your reminder system, you need to change this password. Edit extensions_custom.conf in /etc/asterisk and search for the 123 extension. Change the password entry in the Authenticate entry and reload your dialplan as shown above.

You also can schedule reminders using a web browser. There's an option in FreePBX: Admin, Tools, Reminders. You also can access the reminders application separate and apart from FreePBX using the IP address of your Asterisk server: http://ipaddress/reminders.

The CallerID number for the application, the TTS engine, and your email address all can be adjusted to meet your needs. See the Best of Nerd Vittles article for details on making these changes.

Continue reading Part IV (Monday, May 25).


Twitter Magic. If you haven't noticed the right margin of Nerd Vittles lately, we've added a new link to our Twitter feed. If you explore a little, you'll discover that the user interface now brings you instant access to every Twitter feed from the convenience of the Nerd Vittles desktop. Enjoy!


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...

Asterisk on Steroids: Introducing the Orgasmatron Installer

If an Asterisk® distribution with every bell and whistle on the planet is at the top of your Wish List, then the new Orgasmatron Installer may just be your cup of tea. Let’s face it. The Asterisk learning curve is horrendous. As some of you know, we have built some custom PBX in a Flash systems for the Dell, Everex, and Atom platforms. These builds differ from the PBX in a Flash base install in that they were turnkey PBXs with dozens and dozens of custom applications, extensions, and trunks already preconfigured. While you still needed to change some passwords and plug in some phones, the Orgasmatron builds reduce the Asterisk learning curve to almost zero. Out of the box, email works. Faxing works. ENUM works. Interconnecting Asterisk servers for free calling works. And extensions for 15 phones already are in place. Plug in your Vitelity credentials, and you can place calls to any phone in the world using your new VoIP PBX in a couple of minutes. That’s the good news.

The problem with these builds lies in their basic architecture. To date, all of them were really Mondo backups. And once you strayed from the platform on which the original system was built, your odds of getting a successful restore went down the toilet quickly. Well, that was then. And this is now!

Today we introduce an installation script for PBX in a Flash that lets you build a PBX in a Flash base system, run the Orgasmatron Installer script, and boom! Within a few minutes, you’ve got an Asterisk-based Orgasmatron server on the computer platform of your choice regardless of processor, disk controller, disk drive, network card, and video adapter. And it works equally well in a virtual environment using an open source platform such as the fantastic and free Proxmox Virtual Environment.

Update: Be sure to check out the latest Orgasmatron V Installer at this link.

For those that are wondering what’s included in this new Orgasmatron build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with Asterisk 1.4 or 1.6, FreePBX 2.5, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin:

Getting Started. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. 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 precautions to protect your phone bill. Start by reading our Primer on Asterisk Security. Then 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.

Installation. Here’s a quick tutorial to get you started. First, install the 32-bit version of PBX in a Flash with Asterisk 1.4. Boot your system from the installation CD and type ksalt to begin. When your machine reboots, remove the CD and choose option A to load the most stable payload. When the install completes, reboot your system once again and login as root with the password you chose when you built your system. Now issue the following commands to bring your system current and protect your system passwords: update-scripts, update-fixes, passwd-master. 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.

Now you’re ready to run the Orgasmatron Installer. While still logged into your new server as root, issue the following commands:

cd /root
wget http://pbxinaflash.net/orgasmatron/orgasmatron.x
chmod +x orgasmatron.x
./orgasmatron.x
reboot

Stick around while the install script is running. Parts of it are interactive. For now, choose the Flite option when you’re prompted for text-to-speech preferences. That way you’ll have a working system when you’re finished. Once the installer script is finished, type status and write down the IP address of your server. You’ll need it in the next step to log into FreePBX.

Using a web browser, open FreePBX on your new server with a command like this (substituting the IP address you wrote down above). When prompted for your account name, type maint and use the password you assigned when running passwd-master above:

http://192.168.0.123/admin/

You’re NOT done yet!

These next four steps are important. They get all of the FreePBX modules installed and then restore the FreePBX backup set that’s at the heart of the Orgasmatron build. Just follow along here, and don’t skip any steps. It’s easy.

1. Choose Module Admin, Check for Updates online, Upgrade All, Process, Confirm, Return, Apply Config Changes, Continue.

2. Choose Module Admin, Check for Updates online, Download All, Process, Confirm, Return, Apply Config Changes, Continue.

3. Repeat the above #2 commands a second time.

4. Click on the Tools tab and choose Backup & Restore, Restore, RightNow, and select the .tar.gz file that is displayed. Then choose Restore Entire Backup Set, OK, Apply Config Changes, and Continue.

Securing Your System. You’re almost done. We always like to reboot the server just to make sure nothing got lost in the shuffle. When the reboot is finished, log into FreePBX with a browser again. Before you do anything else, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit after changing secret and Voicemail Password. Repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now let’s change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue. Whew! Your system now is relatively secure. Follow the steps in the tutorials we recommended, and you’re ready to experiment. Plug in a SIP phone or softphone and configure it using one of the available extensions together with the secret for that extension.

Finally, be sure to change the credentials on all of your trunks to match those assigned by your providers. And, in the case of the remote-peer trunk, change the secret and IP address to match the identity on your host Asterisk server. If you don’t have another Asterisk server, change the password anyway so no one can break into your system. Better yet, just delete the trunk unless you plan to use it down the road. We’ll have more to say about this next week. For now, just make up your own, secure password to protect this trunk from outside access by unwanted visitors.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then next week, we’ll cover in detail how to customize every application that’s been loaded. For openers, we recommend you set up an account with Vitelity using our special link below. This gives your PBX a way to communicate with every telephone in the world, and it also gets you a 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 VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. After you’ve connected a phone to your new system, begin your adventure by dialing these 10 numbers:

  • D-E-M-O – Check out the Nerd Vittles Orgasmatron Demo
  • 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 a regular or recurring phone reminder
  • Dial *68 – Schedule a hotel-style wakeup call on any extension

Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. Then log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. Enjoy!

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV (Monday, May 25).


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…

A Baker’s Dozen Asterisk Nuggets from the Forums

Whether you’re new to the Open Source VoIP Community or an old-timer, we wonder how many folks actually miss many of the terrific Asterisk® applications that are hidden in message threads on the various Asterisk forums around the globe. In honor of St. Patty’s Day, today we want to take a stroll through the PBX in a Flash forum just to demonstrate what you may be missing by not visiting the forums or subscribing to some of the better syndication feeds.

Skype Gateway to Asterisk. If you read our recent column on integrating a Skype gateway into your Asterisk server, good for you. But, if you missed the forum dialog which followed release of the article, you missed all sorts of enhancements and system integration tips which made the Skype gateway a much better fit on many systems.

CallerID Superfecta. One of the most perplexing issues facing those that implement VoIP telephony solutions is wrestling with CallerID issues which flow from the ongoing Baby Bell phonebook monopoly. Many of you may have tried our CallerID Superfecta application which provides CallerID lookups for FreePBX-based systems using AsteriDex, Google Phonebook, AnyWho, and WhitePages. But, if you’d explored the forum additions to CallerID Superfecta, you would have uncovered an incredibly slick FreePBX installer as well as support for WhoCalled.us and Telcodata plus SugarCRM as well as numerous fixes for syntax changes on the various lookup sources.

Faxing with Asterisk. Other than CalleriD, there’s probably no issue that generates more consternation in the Asterisk community than fax integration. We reintroduced nvfax for Asterisk 1.4 recently. But, if you’d been following the forums, you’d also know that HylaFax and AvantFax now can be easily integrated into PBX in a Flash thanks to the work of Joe Roper and Tony Shiffer.

A2Billing for Asterisk. Another application that’s been difficult to get working with Asterisk has been A2Billing, a sophisticated calling card and PBX billing system. There really never has been a clear, concise cookbook for getting the software installed and properly configured. Once again, thanks to Joe and Tony, this forum thread provides a step-by-step tutorial for getting every facet of A2Billing installed and properly configured.

Asterisk Stickies. This is another promising Asterisk web application for PBX in a Flash that pops up stickies when incoming calls are received. You then can add the contact to your phonebook and also generate the XML code to update the phone directory on Grandstream and Cisco phone sets. It also supports click-to-dial from the web interface. You can keep up with the progress of this developing application in this very active message thread.

Text-to-Speech FreePBX Module. Just today a new TTS module for FreePBX was introduced which lets you generate TTS announcements for use with any FreePBX-based Asterisk system.

Overhead Paging with Asterisk 1.4. Most workplaces need some sort of overhead paging system. With the tips in this thread and any Asterisk 1.4 server, it’s incredibly easy to implement.

Streaming Music on Hold. We introduced streaming audio for Asterisk over three years ago in the Asterisk 1.2 days. A new message thread has updated that technology to support Internet radio using any Asterisk 1.4 server.

Email Alerts on Trunk Failures. For those that rely upon Asterisk systems to do real work, it’s essential to know when access to your carrier has failed so that you can make adjustments to your outbound and inbound trunks. This thread provides a simple tutorial and script to get you started.

Outbound Emails with Asterisk and SendMail. Another one of our Top 5 most perplexing problems with Asterisk is getting an outbound email capability with SendMail working reliably. Part of this is the configuration hassles with SendMail. But service providers such as Comcast have made matters worse by blocking outbound access to port 25 on most non-business accounts. Here’s a message thread that will walk you through configuring SendMail to use Gmail as your outbound SMTP relay host, and you’ll never have an email problem again on your Asterisk server.

Voicemail Notification. Unified messaging may be everyone’s dream but the reality is that it would be nice to be called on your cellphone when a new voicemail arrived at your office. The Voicemail Notification System does just that. And this thread integrates the original design into a FreePBX module.

Configuration Editor for FreePBX. FreePBX stores much of its magic in Asterisk config files. At least in PBX in a Flash, we hide some of these files to protect the integrity of your system. In addition, changes made to some of these files will get overwritten the next time FreePBX is started since it populates a lot of the information in these config files from data stored in MySQL tables. For those that want to learn more about the FreePBX, there now is a configuration file editor which will let you view and edit any FreePBX config file on your system. You’ll find a complete tutorial in the forums.

Hotel-Style Wakeup Calls. A few weeks ago we covered Tony Shiffer’s new add-on module for FreePBX that provides hotel-style wakeup calls for Asterisk systems. This code actually had been available in the forum for several months and is yet another reason to frequently check the new message threads.

Mac OS X Scripting Package. Since publication, a new link to a Treasure Trove of Goodies for Mac OS X has been posted including a link to the new Mac OS X Scripting Package and Asterisk binaries for Mac OS X from Sven Slezak at Mezzo.

Syndication Syntax. Many forums provide a syndication feed link, but many do not. For vBulletin-based forums, the basic syntax for an RSS feed looks like this:

http://fqdn.com/forum/external.php

You can refine the type of feed you want by specifying the type: RSS, RSS2, ATOM, or XML. For example, to pull down a feed from the PBX in a Flash forum, here’s the syntax for the various formats that are supported:

http://pbxinaflash.com/forum/external.php?type=RSS

http://pbxinaflash.com/forum/external.php?type=RSS2

http://pbxinaflash.com/forum/external.php?type=ATOM

http://pbxinaflash.com/forum/external.php?type=XML

You can further refine the feed by narrowing it down to a particular forum of interest. For example, to retrieve the latest threads from the PBX in a Flash Open Discussion forum, the syntax looks like this:

http://pbxinaflash.com/forum/external.php?type=RSS2&forumids=2

Finally, here’s the list of forum ID numbers for the PBX in a Flash forum:

2 – Open Discussion
3 – Help
4 – Endpoints
5 – Trunks
6 – Providers
7 – Wish List
9 – Bug Reporting & Fixes
10 – Add-On Install Instructions

Something We Missed? There are hundreds of additional Asterisk apps hiding in the woodwork. Please share your discoveries by posting a comment and link below. Enjoy!


Want a Bootable PBX in a Flash Drive? Our Atomic Flash bootable USB flash installer for PBX in a Flash has been quite the hit. Special thanks to all of our generous contributors! Atomic Flash provides all of the goodies in the VPN in a Flash system featured last month on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and this USB flash installer in less than 15 minutes!

If you’d like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment with your best PBX in a Flash story.1

Be sure to include your real email address which will not be posted. The winner will be chosen by drawing an email address out of a hat (the old fashioned way!) from all of the comments posted over the next several weeks.

And it still isn’t too late to make a contribution of $50 or more to the PBX in a Flash project and get a free Atomic Flash installer delivered to your door as our special thank you gift. See this Nerd Vittles article for details.


 

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. This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. []

The Lean, Mean Asterisk Machine: And Now It’s a Fax Machine

Hard to believe it’s been a year since PBX in a Flash hit the street, but today’s the Big Day! So Happy Birthday to us. With an estimated 100,000 downloads worldwide and over a million RSS feeds to our Kennonsoft User Interface each month, you might be wondering what keeps us going with all the reported venture capital behind Big Orange and Lime Green. Well, we’re glad you asked. Truth be told, it’s the cushy offices (in our kitchens) and the endless flow of generous contributions from grateful users. Heh, heh! Seriously, there are some real reasons that account for the popularity of PBX in a Flash. Bottom Line: It Just Works! And here’s a representative sample of other feedback from our fans:

  • Currency – The PBX in a Flash distribution is always up to date. Our separate payload file makes it easy. No one else has anything close. So their builds are almost always long in the tooth.
  • Upgradability – Unlike the competition, you don’t have to start all over each time a new version of Asterisk® or Linux hits the street. We’ll have more to say about our new SUSHI (Software Update Service – Hyperlinked, Interactive) in coming weeks.
  • FlexibilityPBX in a Flash remains the only distribution that builds Asterisk from source. Even Digium®’s own distribution now uses RPMs. When you add new hardware or upgrade the Linux kernel to plug a security vulnerability, you’ll understand why this is critically important.
  • SupportPBX in a Flash has the best support group in the business. It’s called the PBX in a Flash Forum, and it’s free. Unlike the competition, you don’t have to pay to get help on basic technical issues with our product. And you don’t normally wait more than an hour or two for a response. That’s what Open Source is all about!
  • Security – We take security seriously. It’s our number one priority. When there’s a known problem, we don’t hide it or ignore it. We fix it right now. And the RSS Feed that’s part of our KennonSoft User Interface lets you know about it immediately. You can make your own comparisons and draw your own conclusions with regard to the other distributions.
  • No Slimeware – We’re up front about the way we operate and why. We don’t create backdoors or Trojan Horses in our distribution that phone home for any reason. We notify users of issues through an RSS Feed. We believe it’s up to you, not Big Brother, to decide whether to protect your own system. As permitted by the GPL, we do encrypt some of our freeware installation scripts because of the conduct of some in this business that pass off the work product of others as their own.
  • No Bugs – People chuckled when we began a year ago with this mantra because of the experience we all had in days of old. We still believe it and do our best to keep the PBX in a Flash distribution bug free. If you don’t believe it, visit our forums and then visit the others. Some bugs obviously are beyond our control, but we do endeavor to steer users toward stable versions of open source products that can be used reliably in almost any business environment.

So there’s a quick update on how we’re doing and why we do things the way we do. Unlike a year ago, there are lots of choices now in the marketplace. If you’re still on the fence, the nice part of the open source movement is that it doesn’t cost you anything to try several flavors and make your own decision. Ultimately, we think you’ll choose PBX in a Flash for all of the reasons we’ve mentioned.

2011 Update: This article has been updated to support Asterisk 1.8 using HylaFax, AvantFax, and IAXmodem. Click here for the latest article.

Welcome Back Faxing. That brings us to today’s topic: adding a fax machine to your PBX in a Flash system. With all the distributions, there have been numerous fax options. And the one word that describes most of them is P-A-I-N-F-U-L. We’ve been searching for a way to return to the good ol’ Asterisk@Home days with NVfax. It just worked. Well, today it works again with PBX in a Flash and Asterisk 1.4. And, yes, it should work on the other distributions as well. I’ve had mixed emotions about whether to protect the install script, but I’ve chosen to release it in unencrypted format because I think we all can benefit from the contributions of others while still giving credit to those that contribute. And, yes, I know there’s a difference of opinion about this… for some very good reasons. But the Nerd Vittles contribution to VoIP technology has always been distribution agnostic, and we’ve decided to keep it that way. We’re equally delighted that Philippe Lindheimer has left the hooks in FreePBX to support NVfax so, once you complete this install, you can manage incoming fax calls from the comfort of the FreePBX user interface… even in distributions which no longer call it FreePBX. Ever wonder why these folks didn’t also rename Asterisk while they were in the lobotomy business?

How It Works. There are two pieces to the new faxing mechanism. For inbound faxing, you simply set FreePBX to use NVfax to listen for a fax tone on inbound trunks. We’ve found that 5 is the magic number for detecting a fax tone on most inbound calls. YMMV! You also can dial local extension 329 (F-A-X) and the extension will listen for an incoming fax. In either instance, if a fax tone is detected, the call is routed to a fax context that converts the incoming fax to a PDF document which is then sent to your email address specified in your Fax Handling setup for each Inbound Route on your system. The correct answers for Fax Handling are Fax Extension: System, Fax Email: any email address that works, Fax Detection Type: NVFax, and Pause After Answer: 5. Don’t forget to also enter the Fax Machine Settings under the Setup->General Settings tab in FreePBX. For outbound faxing, we can’t recall this ever working with NVfax, but it does now. Here’s how to set things up. Create a PDF document of anything you wish to send by fax. Name the document so that it corresponds with the phone number of the fax destination, e.g. 6789991234.pdf would mean you plan to send the PDF document to a fax device at the following phone number: 678-999-1234. Now place the document in the /tmp directory on your server. Next, pick up a phone on your system and dial 32948 (F-A-X-I-T). When prompted for the destination fax phone number, key in 6789991234. Once you receive an acknowledgment that your fax has been sent, hang up. It doesn’t get much easier than that.

Prerequisites. Well, there are lots of them. But a stock installation of Asterisk with CentOS works great so long as you also have outbound emailing working and you’ve installed a text-to-speech engine. Either Flite or Cepstral works just fine. All of the bundled distributions should suffice. We actually only use TTS to generate the voice prompts for the outbound faxing so, if you don’t need that functionality, no TTS engine is required. If you need help with outbound emailing, see our PBX in a Flash knol. There also are setup instructions for Gmail and Comcast in the PBX in a Flash forum.

Installing the Fax Software. We’ve written a script which handles all of the heavy lifting for you. Just log into your server as root and issue the following commands:

cd /root
wget http://pbxinaflash.net/source/fax/fax.pbx
chmod +x fax.pbx
./fax.pbx

In less than a minute, you should be all set.

Configuring the Fax Software. First, edit the [faxit] context in /etc/asterisk/extensions_custom.conf to plug in your actual fax number to be displayed on outbound faxes. It should be the 17th line up from the bottom of the file. Save your changes and reload Asterisk: amportal restart. Now load FreePBX using your favorite browser and make the Fax Machine entries in Setup->General Settings. Remember that your return email address must match your server domain name that you set up in /etc/hosts to get outbound email flowing, e.g. pbx.dyndns.org. Next, for each of your Inbound Routes in which you wish to enable fax detection, edit the entry and fill in the Fax Handling options we previously mentioned. To repeat, the correct answers are Fax Extension: System, Fax Email: any email address that works, Fax Detection Type: NVFax, and Pause After Answer: 5. Finally, add Misc Destinations for Fax (329) and FaxIt (32948). Reload your dialplan, and you should be ready to go.

Testing Things Out. The easiest way to assure that your system is properly configured is to attach a real fax machine to an FXS device on your system. Then send a fax to extension 329 (F-A-X). You should receive the fax via email shortly thereafter. That’s only half the battle unfortunately. If you want to receive faxes from outside your PBX, you also need to find a VoIP provider that properly supports faxing. Suffice it to say, all VoIP providers are not created equal when it comes to fax support. Our Best of Nerd Vittles article on faxing will provide some suggestions as well as a few tips and tricks. If you have a standard POTS line connected to an FXO device on your Asterisk server, that’s an even better option. Just make certain that fax detection is enabled on the inbound route for that line.

Don’t be misled by the brevity of this article. It in no way is a measure of the effort that it’s taken to make NVfax work again. One way that you can show your appreciation for the good deeds of others is through the Donate link at the top of our page. There’s no obligation, of course, but it does keep the Little Mrs. from regularly asking, "Tell me again why you do this?" Enjoy and thanks in advance.


Getting Started with PBX in a Flash. There’s a great deal of literature on PBX in a Flash that is yours for the taking. But we wanted to mention a terrific new series of articles in Mark Berry’s blog that are especially well suited for those just learning about VoIP. Have a look. We think you’ll agree.


 

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…