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The Most Versatile VoIP Provider: FREE PORTING



JUST RELEASED: Visit the Incredible PBX Wiki

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

Saturday, June 25, 2011 / 18 Comments on 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.

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

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

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

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  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 PBX: Adding Google Calendars with Asterisk 1.8

Monday, May 23, 2011 / 2 Comments on Incredible PBX: Adding Google Calendars with Asterisk 1.8

One of the exciting new features in Asterisk 1.8 is external calendar integration including iCal, caldav, Microsoft Exchange, and ews. Hats off to Terry Wilson at Digium® for his great work on the calendaring API. Unfortunately, there hasn’t been much written about integrating the calendars into FreePBX®-based Asterisk® systems… until now. Because we’re all about free, our focus today will be Google Calendar integration, but we’ll provide you some links to get any of the other calendar types integrated into PBX in a Flash and Incredible PBX if that’s what floats your boat. In addition to being free, Google Calendar is one of the best calendaring apps on the planet. Its beauty lies in the incredible flexibility it provides for group scheduling and the sharing and transparent integration of dozens of calendars (both public and private) not to mention real-time updating with your favorite Android smartphones. Apple does much the same thing with iCal on the Mac and iPhone… for an annual fee.

The rainbow of colors you see in the left column of the screenshot above each represent a different calendar. All of the calendar entries from all of these calendars are integrated into the single calendar display on the right with a simple color scheme to identify where each calendar entry originated. As mentioned, these calendars can not only be from your friends and business associates, they also can be publicly available, read-only calendars for subject matter such as holidays, schedules of your favorite sports teams and rock bands, school calendars, etc. Take a look at iCalWorld to get a quick handle on what’s available. See also this Google resource.

The one bit of advice we would provide in setting up Google Calendar is don’t put all your eggs in one basket. Create separate calendars for different individuals and for different types of events. And don’t mix business with personal. Also keep internal business events separate from those that you might want to share with customers. Down the road, this facilitates sharing just the information of importance to different groups of friends and business associates. Bottom Line: Spend a little time thinking about organization of your calendars. It will pay off handsomely down the road.

Once you integrate Google Calendar into Incredible PBX, you can use it to alert you to upcoming appointments, to schedule and activate conference calls, and even to log the date, time, length, and recipient of all your outbound and/or incoming phone calls.

For today, we’ll get the software installed and functioning. And we’ll help you set up a simple reminder system based upon appointment entries in your Google Calendar. We’ll also give you some good reference materials so that you can experiment a bit on your own. In coming weeks, we’ll build a time slips system to keep track of all your phone calls.

Prerequisites. As mentioned, you’ll need a Gmail account as well as Asterisk 1.8 and PIAF-Purple with Incredible PBX 1.8 to follow along in today’s tutorial. All of the pieces are optional except for Asterisk 1.8 and a fairly current version of FreePBX, but we’ll leave that exercise for the pioneers. The script is licensed as GPL code so knock yourself out.

The Gotchas. Nothing is ever easy, is it? There always are a few surprises to worry about. Here’s the current short list for Asterisk Calendaring. When you schedule a meeting or appointment with Google Calendar, there is an option to set an alert to remind you of the meeting at any time you like on a per appointment basis. So, if you have a meeting 50 miles from home, you could set the alert for 90 minutes in advance while a meeting next door could be set with a 5 minute alert. All of that works fine in Google Calendar. Unfortunately, in Asterisk 1.8.4, there’s a math bug that subtracts the hour differential between your time zone and Greenwich Mean Time (GMT) from these alert times. If you happen to live in the GMT time zone, there’s no bug. However, those living on the East Coast of the U.S. will find their reminder calls arriving four hours too early. If you set an alert for 60 minutes before a scheduled meeting, Asterisk will call you 5 hours in advance. For our West Coast friends, it’s an 8 hour bug. Fortunately, the Asterisk Calendaring config file provides an option to override these alerts with a reminder call at a fixed number of minutes before each appointment. While it lacks the individual appointment flexibility to schedule alerts of different durations, it does work. So, for the time being, we’ll be using that.

Asterisk Calendar alerts also were supposed to be triggered only for events in which your Status has been set to Busy. A typical example would be a medical appointment. Instead, the alerts are generated for all calendar entries. So, if you’ve got 12 children and you’ve set all-day calendar entries to remind you of their birthdays each year, be forewarned that using your default Google Calendar will trigger a friendly reminder call at 11:50 p.m. on the night before each of your kids’ birthdays. Unless you and your spouse are both Night Owls, this might lead to some "WAF issues" if you get our drift. The workaround that we’ll implement today is to create a secondary appointments calendar and reserve it just to schedule meetings and appointments for which you wish to receive reminder calls at 10 minutes before each event. You can change the notification time to any fixed number of minutes desired. As we mentioned, one nice feature of Google Calendar is that entries on secondary calendars will automatically display on your main calendar. Just be sure to schedule the appointments on the correct calendar (we’ll show you how) if you want a reminder call.

Setting Up a Google Appointments Calendar. Before we get Asterisk configured for calendaring, let’s first set up your secondary Google appointments calendar. Log into the Gmail account that you either already use or will begin using for scheduling Calendar entries. Just click on the Calendar tab at the top of the screen and set up your default calendar if it doesn’t already exist. Once it’s set up, there will be a calendar list in the left margin under My Calendars. Just below your existing Calendar(s), there will be a Tasks entry, and below that will be an Add link to add a new calendar. Click on Add and fill in the main pieces of information as shown below. Use appointments for the name of the calendar and make sure you set the correct Time Zone for your hometown. Then save your entries by clicking the Create Calendar button.

Once you have created your new appointments calendar, we need one other piece of information to use it with Asterisk. Under My Calendars in the left margin, click on the down arrow beside appointments and choose Calendar Settings. That will bring up a form that looks like this:

The next to the last section of the form is entitled Calendar Address. You’ll see an entry to the right of the section title called Calender ID with a long alphanumeric string followed by @group.calendar.google.com. Write down the Calendar ID or copy it to clipboard. Don’t include @group.calendar.google.com! You can ignore the statement that the link is inoperative unless you make your calendar public. It actually works just fine with your Google credentials which we’ll configure as part of the Asterisk setup below.

Configuring Asterisk for Google Calendar Integration. PBX in a Flash and Incredible PBX come with most of the components you’ll need to get calendaring to work. However, there are a few missing pieces at the moment. These will require that Asterisk be recompiled, and we’ve built a little script to do all of it for you. Just log into your server as root and issue the following commands:

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

After you enter your Google credentials and Calendar ID, the script will temporarily shut down Asterisk, download the missing components for calendaring, recompile Asterisk, and customize the calendar setup with your Google credentials. Then the script will complete the setup and restart Asterisk. The whole process takes about 5 to 10 minutes.

Asterisk Calendar Configuration. Here are the pieces in the default Google Calendar config file that will be replaced with your personal settings:

[GoogleCalendar]
type = caldav
;url = https://www.google.com/calendar/dav/username@gmail.com/events/
url = https://www.google.com/calendar/dav/CalendarID@group.calendar.google.com/events/
user = username@gmail.com
secret = userpassword
refresh = 10
timeframe = 120
;
autoreminder = 10
;
channel = local/8005551212@from-internal
context = from-internal
extension = 225
waittime = 45

If you should ever need to make changes to these settings again, you’ll find the entries with your Google credentials at the end of the calendar.conf config file in /etc/asterisk. Remember to restart Asterisk after making any changes. The commented out URL entry is the one you’d use to access your main Google Calendar instead of the secondary appointments calendar we created above. If the GMT bug is ever fixed, this would be a step forward.

Extension 225 (C-A-L) is the extension that will be called to actually play back the reminder information. You’ll find this block of code at the top of extensions_custom.conf in the [from-internal-custom] context. You can make any adjustments you’d like. We simply set this up to give you a template to follow. As configured, the reminder will read you a summary, the starting time, location, and any description associated with each calendar entry. If you have a Cepstral voice in addition to Flite, just change the Flite entries to Swift. Then restart Asterisk: amportal restart.

Asterisk Calendar Test Drive. Asterisk keeps track of your scheduled events by checking your Google Calendar every 10 minutes. We recommend configuring reminders for calls 10 minutes before any scheduled event in the appointments calendar. What this means is you probably should never schedule an event expecting to get a reminder call if the event is less than about a half hour in the future. If you use a setting for the autoreminder time other than 10 minutes, this will obviously be different for you.

So let’s schedule an appointment for an hour from the current time. Click on the Calendar tab in Gmail. And then click on the down arrow beside the appointments calendar in the left column. Choose Create Event on this Calendar. You’ll get a form that looks like this. Click Save to store your new Calendar entry. Then go have a beer and wait for your call.

Where To Go From Here. We’ve just scratched the surface of what you can do with Asterisk 1.8 using Google Calendars. We recommend you now review the chapter in the new Asterisk Definitive Guide book that covers calendaring in some detail. It will give you a good handle on what’s possible. For a more technical discussion, take a look at this contribution from one of the PIAF Forum regulars. Amazing stuff! There’s also a good article in VoIP Today that’s worth a careful read. Be aware that, if you recompile Asterisk to add support for Microsoft Exchange, you probably will break the Google Calendar connection that we’ve outlined today. At least, we never have been able to get the two to coexist. If you have better luck, please post your results. The reason for the problem is that the Exchange connector depends upon neon-0.29 which is not yet part of CentOS 5.x. We’ll keep you posted on our progress. Enjoy!

Originally published: Monday, May 23, 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 Try the New, Free PBX in a Flash Conference Bridge.

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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

Some Recent Nerd Vittles Articles of Interest…

Incredible Fax: Free Faxing Returns to Incredible PBX 1.8

Monday, May 2, 2011 / 9 Comments on 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 [↩]

Incredible PBX for Asterisk 1.8: Back from the Brink

Monday, December 13, 2010 / 25 Comments on Incredible PBX for Asterisk 1.8: Back from the Brink

Photo courtesy of Blou_Aap. Artist unknown but terrific.Ever had one of those weeks? It was a wild ride these past 7 days with the introduction of Asterisk® 1.8.1 and some new Google Voice twists. And then there was our DNS provider Omnis.com that trashed name resolution for our primary domain, pbxinaflash.net, while claiming with a straight face that they didn't provide tech support for their own stupidity. Yikes! But where there's a will, there's always a way. And by Friday night, not only were all the issues sorted out but the Google Voice Gtalk interface in Asterisk 1.8 for free calling in the U.S. and Canada is now better than ever. Our special thanks to the Asterisk Dev Team and Tom King of the PIAF Dev Team for restoring peace in the valley. No more callback hoops for outbound calling. Free DIDs in most area codes. Instantaneous connections. Crystal clear calls. You can almost hear a pin drop. And Incredible PBX now brings you all this magic in a turnkey install that even a monkey could handle.

As if we needed another one, our other surprise last week was the Ebay appearance of a Nortel SIP Videophone labeled as a 1535, but it had no WiFi in either the hardware or in the particular software build. The merchant was as surprised as we were to discover the missing WiFi component and now has corrected the ad. But that won't make the WiFi reappear. For those of you still purchasing these phones (and they're worth it), read the fine print if WiFi or firmware upgrades matter to you. The Turkish models have neither. As anyone that tracks Ebay auctions will tell you, the law of supply and demand controls the price. These began in the $30 range and as recently as two weeks ago were selling for almost $80. They've now dropped back into the $50-$60 range. You're usually better off calling the merchant, and the more you buy, the better the price. Five Stars Telecom usually stocks the U.S. models. But ask to be sure.

So here's the drill today. Just download the brand new PBX in a Flash 1.7.5.5.4 ISO with the newly patched Asterisk 1.8 Purple payload. Then burn the ISO to a CD and boot your server from the PIAF CD. Choose the Purple Edition after CentOS installs which will load Asterisk 1.8.1 with FreePBX 2.8. Finally, run through the 5-minute install of Incredible PBX for Asterisk 1.8.1. In less than an hour, you'll have a turnkey, secure 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. Not only can you start enjoying free phone service immediately, but you'll have a robust PBX platform that will keep your eyes popping for months learning about all the features that would have cost you hundreds of thousands of dollars less than a decade ago. Did we mention that all of this telephone goodness is absolutely FREE!

Thanks to its Zero Internet Footprint™ design, The Incredible PBX also 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.

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 with CentOS 5.5, Asterisk 1.8, FreePBX 2.8, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, 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. Here are the 5 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 1.7.5.5.4 Purple Edition
2. Download & run The Incredible PBX 1.8 installer
3. Run passwd-master on your PIAF server
4. Map UDP 5222 on firewall to PIAF server
5. 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, we recommend the very latest 32-bit version of PBX in a Flash 1.7.5.5.4.3 If you installed it last week, that's not new enough. The ISO hasn't changed, but the Purple payload is radically different since this morning! Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.5 operating system. That hasn't changed. But, 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. To get the patched version of Asterisk 1.8.1, use today's new 1.7.5.5.4 ISO. Choose the new Purple Payload, and our special Asterisk 1.8 patched release and all of the Google Voice goodies will be configured automatically. And you won't have to worry about the CDR crashing your new server either.

You can download the 32-bit PIAF 1.7.5.5.4 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 #1: 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! Press Ctrl-C to cancel the install.

WARNING #2: The PIAF Dev Team currently classifies PIAF-Purple and Asterisk 1.8 as E-X-P-E-R-I-M-E-N-T-A-L. Remember the Pioneers! If you have a low threshold for pain, if you depend upon your PBX to actually make and receive phone calls, or if you understand the WAF and prefer sleeping with both eyes closed, abort this install now and choose PIAF-Gold, PIAF-Silver, or PIAF-Bronze. Otherwise, enjoy the ride!

On some systems you may get a notice that CentOS can't find the kickstart file. Just tab to OK and press Enter. Don't change the name or location of the kickstart file! This will get you going. Think of it as a CentOS 'feature'. 🙂 If your system still won't boot, then you have an incompatible drive controller.

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 and update-fixes for you. You still should set your FreePBX passwords by running passwd-master after The Incredible PBX installer finishes!

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

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.

Running The Incredible PBX Installer. 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
passwd-master

If you've installed the previous version of The Incredible PBX, you'll recall that there was a two-step install process after configuring another trunk with either SIPgate or IPkall. That's now a thing of the past. All you need to do after The Incredible PBX script completes is run passwd-master to set up your master password for FreePBX.

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

Google Voice Account Name
Google Voice Password
Google Voice 10-digit Phone Number
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 Google Voice Phone Number is the 10-digit DID for this dedicated account. We need this if we ever need to go back to the return call methodology for outbound calling. For now, it's not necessary. But who knows what the future holds. 🙄 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. You'll actually set it by running passwd-master after The Incredible PBX completes. We need this password to properly configure the CallerID Superfecta for you. By the way, none of this confidential information ever leaves your machine... just in case you were wondering.

Now have another 5-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 (and so you don't forget), go ahead and configure your hardware-based firewall to support Google Voice. See the next section for what's required. Without completing this firewall configuration step, no calls will work! When the installer finishes, READ THE SCREEN just for grins.

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.

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!

Firewall Configuration. We hope you've taken our advice and installed a hardware-based firewall in front of The Incredible PBX. It's your phone bill. You'll need to make one adjustment on the firewall. Map UDP 5222 traffic to the internal IP address of The Incredible PBX. This is the port that Google Voice uses for phone calls and Google chat. You can decipher the IP address of your server by logging into the server as root and typing status.

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. If not, make certain you are not logged into Google Chat on a Gmail account with these same credentials. 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, and 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.

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!

This latest version of Incredible PBX locks down your server to private networks and existing, registered Asterisk devices. Should you need to enable additional IP addresses for other devices or providers at a later date, simply add the new IP addresses to /etc/firewall.whitelist and then rerun /root/firewall-whitelist.sh. For additional background, read this article.

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 just completed. We'll have more details and additional utilities for your use in coming weeks.


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 getting the final kinks out of Jabber with Asterisk. 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 turned Asterisk 1.8.1 into something special. No one will quite understand what an endeavor that was until they try it themselves. And, finally, to our legion 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/nv folder. You'll find all sorts of goodies to keep you busy. 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, December 13, 2010

Quirks & Bugs. Well, there aren't any that we know of. But we'll keep a running list here so you can check back from time to time if you don't participate in the PBX in a Flash Forums.

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Safely Interconnecting Asterisk Servers for Free Calling

Adding Skype 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.


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.

 

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

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

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

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

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

Some Recent Nerd Vittles Articles of Interest...

  1. For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. [↩]
  2. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. 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 [↩]
  3. HINT: Version 1.7.5.5.3 ISO also works just fine. [↩]

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

Friday, December 10, 2010 / 91 Comments on 5 Steps to Paradise: It’s Incredible PBX for Asterisk 1.8

Hard to believe it's been almost six months since we introduced The Incredible PBX, but that makes today even more special. With the release of Asterisk® 1.8, the PBX in a Flash Development Team headed up by Tom King burned the midnight oil to introduce the latest PBX in a Flash Purple Edition with Asterisk 1.8 in less than 24 hours.

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

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

So we had all the tools necessary to reengineer, design and build the all-new Incredible PBX for Asterisk 1.8. What used to be a somewhat 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 now 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 on extensions 701 and 702.

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, 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. Here are the 5 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. Run passwd-master on your PIAF server
4. Map UDP 5222 on firewall to PIAF server
5. 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, you must install the latest 32-bit version of PBX in a Flash.3 Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. That hasn't changed. But, 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! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can't find the kickstart file. Just tab to OK and press Enter. Don't change the name or location of the kickstart file! This will get you going. Think of it as a CentOS 'feature'. 🙂 If your system still won't boot, then you have an incompatible drive controller.

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 and update-fixes for you. You still should set your FreePBX passwords by running passwd-master after The Incredible PBX installer finishes!

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

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.

Running The Incredible PBX Installer. 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
passwd-master

If you've installed the previous version of The Incredible PBX, you'll recall that there was a two-step install process after configuring another trunk with either SIPgate or IPkall. That's now a thing of the past. All you need to do after The Incredible PBX script completes is run passwd-master to set up your master password for FreePBX.

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. You'll actually set it by running passwd-master after The Incredible PBX completes. We need this password to properly configure the CallerID Superfecta for you. 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 (and so you don't forget), go ahead and configure your hardware-based firewall to support Google Voice. See the next section for what's required. Without completing this firewall configuration step, no calls will work! When the installer finishes, READ THE SCREEN just for grins.

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.

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!

Firewall Configuration. We hope you've taken our advice and installed a hardware-based firewall in front of The Incredible PBX. It's your phone bill. You'll need to make one adjustment on the firewall. Map UDP 5222 traffic to the internal IP address of The Incredible PBX. This is the port that Google Voice uses for phone calls and Google chat. You can decipher the IP address of your server by logging into the server as root and typing status.

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, and 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.

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 just completed. We'll have more details and additional utilities for your use in coming weeks. Stay tuned!


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 Asterisk 1.8.0 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/nv folder. You'll find all sorts of goodies to keep you busy. 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, November 1, 2010

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Safely Interconnecting Asterisk Servers for Free Calling

Adding Skype 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.


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

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

 

Special Thanks to Our Generous Sponsors


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

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

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

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

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

Some Recent Nerd Vittles Articles of Interest...

  1. For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. [↩]
  2. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. 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 [↩]
  3. HINT: Version 1.7.5.6 recommended, but 1.7.5.5.3+ ISOs also work just fine. [↩]

Avoiding a $100,000 Phone Bill: VoIP WhiteList for IPtables

Friday, December 3, 2010 / 1 Comment on Avoiding a $100,000 Phone Bill: VoIP WhiteList for IPtables

It’s been almost a year since we last wrestled with VoIP security for Asterisk®. With Christmas just around the corner, it seemed like a fitting time for a report card. Suffice it to say, the bad guys have not stood still. Attacks have become much more frequent and more sophisticated as VoIP systems have proliferated. A year ago we saw brute force attacks with thousands of password attempts on VoIP servers. These attacks could easily be detected by Fail2Ban. What we are seeing today are one and two hit drive-bys that usually are initiated from Windows zombies or hosted accounts established with stolen credit cards. These VoIP attacks fly under the radar unless you review your logs every day. Have the creeps gotten more patient? No, just smarter. They now understand the VoIP security model that has been deployed on systems like PBX in a Flash, and they simply work around it. Two hits per server, and they’re off to the next IP address only to return in a few hours to try two more. Are these attempts successful? Well, here’s the latest recipient of a $100,000 phone bill so the answer would appear to be affirmative.

We continue to wrestle with new security approaches to better protect Asterisk VoIP systems, and we’ve stumbled upon another golden arrow for your security quiver. Our Incredible PBX platform continues to offer the very best security solution because it is designed to sit safely behind a hardware-based firewall with virtually no exposure to the Internet. But such deployments assume that both your server and your phones are all safely ensconced behind a hardware-based firewall. If it turns out that you want to deploy a SIP phone for use by grandma or you’ve decided you’d like to try hosted PBX service from a provider such as rentpbx.com,1 then there either need to be holes opened in the firewall or there is no hardware firewall protection in the case of hosted service.

Over the past few weeks, we’ve explored a number of new security approaches to better protect your Asterisk server. These include The SunshineNetworks Knock as well as VoIP Black Lists and VoIP White Lists. If you’re technically savvy, you’ll want to carefully consider "The Knock" for all of your SIP phones exposed to the Internet.

We spent a good bit of time considering various VoIP BlackList solutions. As the name implies, a list of the bad guys’ IP addresses is fed into IPtables which then blocks access to your server from these addresses. Sounds good, right? One approach with a BlackList is to block all IP addresses from "problem countries." The methodology to implement this solution can be found in this thread on the PIAF Forums. The problem, of course, is identifying the "problem countries." Another option was to implement an IPtables Blacklist based upon the work of the VoIP Blacklist Project. Perhaps ironically, the VoIP Blacklist Project actually blocks the IP addresses of both Nerd Vittles and PBX in a Flash, and emails requesting removal of our IP address were ignored. To save time, the VoIP Blacklist Project employs CIDR Masks which can blacklist hundreds of thousands of IP addresses in one fell swoop. Problem is that a lot of innocent people get caught in the net, and there’s no easy way out without maintaining the blacklist yourself. The final dagger in the black list approach is zombies. Insecure Windows machines have been compromised by the droves worldwide and particularly in the United States. So identifying all of these now-malicious systems is not unlike playing Whack-a-Mole. When you block one of them, six more pop up. So, after giving it the good old college try, our view of VoIP Blacklists should be obvious. No, thanks. There are very real risks that the bad guys can and have poisoned existing blacklists with safe IP addresses, and the number of Windows zombies grows geometrically making it all but impossible to have or maintain a blacklist that affords any real protection.

These results with black lists led us to the conclusion that the only real security mechanism that could protect many VoIP servers today was a VoIP WhiteList for IPtables. As the name implies, we want to identify the IP addresses of every SIP and IAX trunk and extension on your server and then feed those addresses into IPtables so that the only access to VoIP resources on your server is from these addresses. Today’s VoIP WhiteList for IPtables consists of two bash scripts: one queries the MySQL database in which FreePBX stores all of the trunk and extension information for your server and the other populates IPtables with the results of the queries. We would hasten to add that a similar white list is equally important for SSH access to your server although we think it is better to implement an SSH WhiteList on your hardware-based firewall. In this way, you can adjust the SSH white list via web browser while traveling without locking yourself out of your Asterisk server.

Prerequisites. To use today’s VoIP WhiteList for IPtables, you’ll need either a current version of PBX in a Flash or Incredible PBX. Other aggregations will also work provided your system is FreePBX-based (version 2.6 or later), has IPtables already installed and functioning properly, and has an /etc/sysconfig/iptables configuration file that closely matches the stock PBX in a Flash design. We’ll leave it to you to make that call after reviewing the scripts.

VoIP WhiteList Design. We’ve designed the VoIP WhiteList for IPtables to be modular. There’s a firewall-whitelist-gen.sh script which extracts from MySQL the list of IP addresses used by your trunks and extensions. This text-based list is stored in /etc/firewall.whitelist. You can manually add and delete entries from the list once it is populated.You also can rerun the script at any time to generate a fresh catalog of WhiteList IP addresses based upon your current trunk and extension settings. This script also enables access to your server from the public IP address of your server as well as all non-routable IP addresses. Finally, it modifies /etc/sudoers slightly so that Travelin’ Man can be used to add dynamic IP addresses on the fly. We’ll cover that below.

The second script is firewall-whitelist.sh, and it is used to actually implement your new VoIP WhiteList in IPtables. The changes take effect immediately. It also can be run again to update these entries if you manually add or delete IP addresses in /etc/firewall.whitelist. This script always creates a backup copy of your previous /etc/sysconfig/iptables file and names it iptables.timestamp where the timestamp is the date and time of your last update, e.g. iptables.12012010-083841 was created on Dec. 1, 2010 at 08:38:41. If you should ever shoot yourself in the foot, simply copy one of the iptables backup files to /etc/sysconfig/iptables and then restart IPtables: service iptables restart.

WARNINGS: In order to implement the WhiteList, the script removes the existing IPtables entries which permit SIP and IAX access from anywhere using UDP ports 4569 and 5000 to 5082. If you have edited these entries in any way, you’ll need to remove them and restart IPtables before running firewall-whitelist.sh. Otherwise, your more general firewall entries will leave your system vulnerable to access from IP addresses not in your VoIP WhiteList.

If your system is running on a hosted server, you’ll need to make a couple of additions to /etc/sysconfig/iptables and restart IPtables (service iptables restart) before running firewall-whitelist.sh, or you may lock yourself out of your own server. Be sure to add the public IP address of your server, and also add the IP address from which you are making changes to your server. Each entry should look like the following example using your actual IP addresses. And the entries should be added above the COMMIT line in the same section of the iptables file as the existing UDP 10000:20000 ACCEPT entry:

-A INPUT -s 222.222.222.222 -j ACCEPT

Installing the VoIP WhiteList for IPtables. Installation is easy. Just log into your server as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/firewall-whitelist.tar.gz
tar zxvf firewall-whitelist.tar.gz
./firewall-whitelist-gen.sh
./firewall-whitelist.sh

If you installed one of the beta versions of the VoIP WhiteList from the PIAF Forums, then you’ll need to do a little housecleaning before actually running either of the scripts. Just edit /etc/sysconfig/iptables and clean out all of the entries that contain 5000:5082 as well as any entries nearby that include the non-routable IP addresses, e.g. 192.168.0.0. Finally, if there are entries beginning with -A WHITELIST, delete those as well. Then restart IPtables: service iptables restart. Thank you for your testing and feedback!

Deploying Remote SIP Phones. What remains is some method for connecting remote SIP phones with dynamic IP addresses. Our Travelin’ Man application was specifically designed to provide this support although the initial version only opened the necessary IP address for Asterisk access. The latest release also provides the necessary IPtables support. You have two options: either remove the old version and supporting directories under /var/www/travelman or edit the index.php file in each subdirectory you’ve created and make the change shown in this post on the PIAF Forums. Enjoy!


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. We gratefully acknowledge the contributions of rentpbx.com to the PBX in a Flash Development Team. In addition to hosted accounts to test PBX in a Flash in the hosted environment, rentpbx.com also has contributed technical assistance particularly as it relates to our Google Voice-Asterisk integration efforts. [↩]

The Incredible PBX: Safely Interconnecting Asterisk Servers

Monday, November 15, 2010 / 5 Comments on The Incredible PBX: Safely Interconnecting Asterisk Servers


 
WOW! What a couple of weeks it has been. The response to Incredible PBX for Asterisk® 1.8 has been, well, incredible. Just last week, SlickDeals and FatWallet introduced over 50,000 bargain hunters to the beauties of Asterisk and Google Voice using Incredible PBX. They joined our regular 50,000 weekly visitors in discovering what may be the best VoIP calling platform on the planet, free or otherwise.

But we’ve also heard from long-time users of PBX in a Flash: "How can we take advantage of this new Google Voice technology without breaking our existing server?" Well, starting today, it’s easy! We’re going to show you how to interconnect as many Asterisk servers as you like using a simple FreePBX tweak to make free calls using your Incredible PBX. To begin, just set up a second server or virtual machine running Incredible PBX 1.8. Then we’ll walk you through interconnecting it with any other Asterisk server that’s running FreePBX. It really is a 5 minute project… once you’ve finished reading this article.

Don’t be intimidated by all of the screen shots shown below. We’re just showing multiple ways of doing the same thing. So you don’t need to use all of them. Once you’ve added one trunk entry on each of your servers and an outbound route on your existing Asterisk server, all of the users on your primary server can instantly begin making free outbound calls through the Google Voice setup on your Incredible PBX. Keep in mind that, at least for now, there is no limit to the number of simultaneous (free) outbound calls you can make within the U.S. and Canada using the Incredible PBX 1.8 platform. And you can interconnect as many Asterisk servers as you like assuming you have the 100kbps VoIP bandwidth to support each simultaneous call.

To get started, follow our last article to get an Incredible PBX 1.8 server set up. As shown in the diagram above, we’re going to assume you’ve got both your new and old Asterisk servers running on the same subnet behind a very secure hardware-based firewall. But this isn’t really required from a technical standpoint. One or more additional servers could be strung all around the globe if that’s your requirement. Or you may wish to take advantage of the incredible deal at RentPBX.com and let them host Incredible PBX 1.8 for you at $15 a month. Just use this special coupon code: BACK10. Then all of your other Asterisk servers can take advantage of today’s free-calling solution. We would hasten to add that, once you’re using the Internet as the transport mechanism for interconnecting servers, we recommend you read and use the secure VPN setup outlined in our VPN in a Flash knol, but the IAX setup outlined below is secure except your voice data is not encrypted. So that’s your call to make.

Today’s Drill. We’re going to show you how to make calls from your existing Asterisk server through The Incredible PBX today. We’ll leave it to you to get things working in the other direction if that is a requirement for your project. First, we’ll create a new trunk on The Incredible PBX, and then we’ll create both a new trunk and a new outbound route on your existing server. We’ll also cover two different interconnection setups. First, we’ll do it using SIP. And then we’ll show you a similar setup using Asterisk’s IAX.

If both servers are sitting on the same private LAN, then the SIP setup is a little easier because the Linux firewall running on Incredible PBX allows SIP traffic to flow freely without any adjustment. It assumes you have added the recommended hardware firewall layer of protection with SIP access to your servers closed off. If one or more of your servers are outside the hardware firewall that is protecting Incredible PBX 1.8, then we recommend the VPN solution referenced above first and the IAX solution outlined here as a second option because the data is unencrypted. Both of these options avoid having to open up any SIP ports on your hardware firewall, and require only a minor adjustment to IPtables, the Linux-based firewall running on The Incredible PBX.

Naming Conventions. To keep things simple, we’re going to refer to the two servers in our example as incredible-pbx and piaf-main where incredible-pbx is your new Incredible PBX 1.8 server that will host the outbound Google Voice calls for users on your piaf-main server. You can obviously adjust these names in any way you like. The only gotcha is that Asterisk attempts to match an incoming call’s username against one of its corresponding trunk names before allowing the call. If there’s no match, the call will fail. So make sure that, if you change the names in the example, do it for both the username and trunk name entries on both servers. Better yet, follow the naming convention in our example, and it just works. 😉

Security Implications. If any of your Asterisk servers allow direct SIP traffic from the Internet, then you need to be extra careful in setting up this interconnectivity since it may allow anyone to attempt to make calls through your Incredible PBX depending upon how your primary server’s dialplan is configured. For example, once a server is interconnected with Incredible PBX, anyone could dial 6789876543@youripaddress and the call might be processed by Google Voice. To avoid this, the simple solution is to password-protect every Outbound Route on your Incredible PBX by adding a Route Password. Or, better yet, don’t expose any of your Asterisk servers to Internet SIP access. Whatever you do, be sure to test making a SIP URI call such as the one shown here once you have all of the pieces in place. Then you’ll know whether you have a security issue or not.

Setting Up Incredible PBX for Interconnecting Servers. Let’s set up a SIP and IAX trunk on your Incredible PBX first. You really don’t need both of these. To repeat, if The Incredible PBX is located on the same private subnet as your other Asterisk server, just use the SIP trunk. If you need access from an Asterisk server outside your private LAN, use the IAX setup. To begin, login to FreePBX using maint and the password you set up with passwd-master. To create a trunk, first choose Setup, Trunks.

To create a SIP trunk, click Add SIP Trunk. For the Trunk Name, enter piaf-main. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.50 with the actual IP address of your piaf-main server. Replace password with a very secure alphanumeric password. Leave the other entries as they are.


 
To create an IAX trunk, click Add IAX2 Trunk. For the Trunk Name, enter piaf-main. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.50 with the actual IP address of your piaf-main server. Replace password with a very secure alphanumeric password. Leave the other entries as they are.

With either or both trunks, you have the option of tightening up how calls placed from the other server are routed. To force all calls to go out through the Google Voice trunk, just change context=from-internal to context=gvoice. If you want extensions on the other server to be able to call extensions on The Incredible PBX directly, leave the context entry the way it is shown.

While we don’t recommend it, if you’re going to have multiple Asterisk servers connecting to The Incredible PBX to place Google Voice calls and you’re too lazy to create separate trunks to support each server, you can eliminate the IP address checking mechanism in Asterisk by replacing host=192.168.0.50 with insecure=port,invite. The security implications should be obvious.

Setting Up The Other Asterisk Server. There are two steps in setting up any other server that you wish to interconnect with The Incredible PBX. First, you have to create a compatible trunk to handle the calls. Then we’ll add an Outbound Route to send certain calls to Incredible PBX for processing. If you’re using SIP on the Incredible PBX, then you have to use SIP on the other Asterisk server. Same goes for IAX. We’ll set up both a SIP and IAX trunk on the PIAF main server just to show you what the entries should look like. And, to repeat, you really don’t need both of these. If your other Asterisk server is located on the same private subnet as Incredible PBX, use the SIP trunk. If you need access to Incredible PBX from elsewhere, use the IAX setup. To begin, login to FreePBX on your other PIAF server using maint and the password you set up with passwd-master. To create a trunk, first choose Setup, Trunks.

To create a SIP trunk, click Add SIP Trunk. For the Trunk Name, enter incredible-pbx. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.212 with the actual IP address of your incredible-pbx server. Replace password with the same secure alphanumeric password you used on the Incredible PBX SIP trunk to which you will be connecting. Leave the other entries as they are.


 
To create an IAX trunk, click Add IAX2 Trunk. For the Trunk Name, enter incredible-pbx. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.212 with the actual IP address of your incredible-pbx server. Replace password with the same secure alphanumeric password you used on the Incredible PBX IAX trunk to which you will be connecting. Leave the other entries as they are.

You’ll notice in the Dial Rules, we’ve used 48 (which is GV on a phone) as the prefix to be dialed on your other Asterisk server to route calls out through Google Voice on The Incredible PBX. So, to place a call from your other Asterisk server via Google Voice, a user would dial something like this: 48-678-987-6543. Before the call leaves the Asterisk server, the 48 prefix will be stripped off. You can make this prefix anything you’d like. Just be sure to use the same prefix when you set up the Outbound Route in the next step.

Adding an Outbound Route. The final configuration step is to add a new outbound route on your other Asterisk server to actually send calls to The Incredible PBX. As noted, we use a dialing prefix so that we can identify the calls to be sent. Create a new route called GoogleVoice and make your entries look like the following if you’re using IAX. If you’re using SIP, just change Trunk Sequence 0 to SIP/incredible-pbx. Click Submit Change and reload FreePBX when prompted.


 

Keep in mind that FreePBX processes Outbound Routes in top down order, and the first matching route is the only route that is used to place the call even if the call fails. So the trick here is to move your new GoogleVoice route up the list so that it’s at least above the default calling route (which is a route with no specified dial patterns to match) and any other routes consisting of 12 or 13-digit dial strings which might match our GoogleVoice dial patterns.

IAX Firewall Adjustments. If you’re using the IAX method above, you’ll need to adjust the IPtables firewall rules on Incredible PBX to allow communications with your other Asterisk server. If your other Asterisk server is PBX in a Flash, you may need to add a similar entry in the IPtables rules on that machine as well. In addition, you’ll need to map UDP 4569 on your hardware-based firewall to the private IP address of your Asterisk server. Otherwise, calls will never make it past your firewall.

On each server, edit /etc/sysconfig/iptables and add an entry with the IP address of the other server with which you’ll be communicating. If your Incredible PBX is on a different public network than your other server, we’d need to add an entry near the end of the file and above COMMIT allowing IAX communications with the public (not private!) IP address of the piaf-main server assuming that server is outside the LAN, e.g. something like this:

-A INPUT -p udp -m udp -s 222.68.100.150 –dport 4569 -j ACCEPT

If you’re using IAX and both servers are on the same private subnet or interconnected private subnets, then the entry might look like this:

-A INPUT -p udp -m udp -s 192.168.0.50 –dport 4569 -j ACCEPT

Once you’ve saved your change, restart the firewall: service iptables restart

Testing Things Out. Now you’re ready to place a test call. Pick up an extension on your piaf-main system and dial 48-800-322-7300. You’ll be greeted by American Airlines courtesy of Google Voice. The CallerID of your outbound calls will be your Google Voice number regardless of the extension or server from which the call originates. Enjoy!

Originally published: Monday, November 15, 2010

Introducing The Incredible PBX for Asterisk 1.8

Adding Skype 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

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.


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…

It’s TeleYapper 5.0: The Ultimate RoboDialer for Asterisk

Sunday, October 10, 2010 / 21 Comments on It’s TeleYapper 5.0: The Ultimate RoboDialer for Asterisk

We don't normally take a month off at Nerd Vittles which should tell you something about today's 10/10/10 column. We're pleased to introduce TeleYapper 5.0, a completely rewritten, Asterisk® 1.4 and 1.6.2-compatible version of our telephone broadcasting service.1 Using Cepstral text-to-speech, TeleYapper 5.0 brings individualized, text-based messaging and customized reminders coupled with the ability to capture recorded responses from every call.

WARNING: Because of changes in Cepstral, this application now requires an additional $200 license from Cepstral. We no longer recommend Cepstral for obvious reasons and will have a comparable system using Google's new Speech-to-Text application soon. Our apologies.

As part of the message delivery process, you now can customize and capture any one of four different responses from those that are called. And TeleYapper 5.0 will email you a CSV and/or XML file with the RoboResponse™ results when the calling process is completed including a list of failed calls and calls that were answered by an answering machine. In addition, you can have TeleYapper email certain call results to various individuals as the calls are processed if your requirements demand it.

For those with multiple outbound trunks, TeleYapper 5.0 supports simultaneous calls using multiple trunks. And now there are significant enhancements that detect answering machines and real people. This lets you deliver customized messages depending upon whether an actual human answers the phone.

Version 5 has been tested extensively with the Gold, Silver, and Bronze editions of PBX in a Flash 1.7.5.5, which provides support for the latest and greatest versions of Asterisk 1.4 and 1.6.2. And it should work well with other Asterisk aggregations with MySQL, Cepstral TTS support, and FreePBX 2.5 or later.

Overview. For those that have never used TeleYapper, here's a quick summary of how the new version works. It's an automated message broadcasting service commonly known as a call blasting or phone blasting system. In addition to loads of creepy uses, phone blasting has legitimate purposes as well. TeleYapper is licensed in several different ways for the following purposes: prerecorded phone messages for neighborhood association announcements, medical appointment reminders, school closings, tornado alerts, little league practices, municipal government reminders. It's free to use for non-profit, civic, and non-political purposes provided you don't solicit money or seek to sway someone's opinion or encourage a particular vote on an issue or candidate. All other uses require a commercial license. For commercial, political, and medical applications, please review our licensing terms below.

How it Works. Step #1 is to create a CSV or XML export from your favorite database application with the information that will be used to send the messages or reminders. This could be as simple as a list of phone numbers or as complex as a listing of doctors and patients with the dates, times, and places of their next medical appointments together with special patient instructions for activity preceding their visit, e.g. "Please remember to start flossing a month before your next dental appointment."

Step #2 is to create a config file with the robodial settings as well as the text which will actually be spoken during each customized call. If you remember form letters from your word processing days, TeleYapper's config file offers the same flexibility. A message can be as simple as "Take cover immediately. A tornado has been spotted at the end of your street." Or it could be a medical appointment reminder such as the following:

Hi. This is Allison from Charleston Family Clinic calling to confirm Jan's appointment with Doctor Quack on Tuesday, October 5th, 2010, at 10:30 a.m. in our Charleston office. Please remember not to eat or drink anything after midnight on the night before your scheduled appointment.

To confirm your appointment, press 1. To reschedule your appointment, press 2. To cancel your appointment, press 3. If we have reached you in error or if you do not wish to receive further automated medical appointment reminders, press 4. To hear this message again, please press 5 now.

And you can create a separate message which would be delivered in the event an answering machine takes the call:

Hi. This is Allison from Charleston Family Clinic calling to confirm Jan's appointment with Doctor Quack on Tuesday, October 5th, 2010, at 10:30 a.m. in our Charleston office. Please remember not to eat or drink anything after midnight on the night before your scheduled appointment.

If you need to change or cancel your appointment or if we have reached you in error, please call our office at your earliest convenience. The number is 800-123-4567. Goodbye.

Step #3 is to use your web browser to access a password-protected web page that will let you upload your CSV or XML data and your config file to kick off the dialing spree. Once the files have been uploaded, everything else is automatic.

Step #4 is to sit back and relax while TeleYapper executes your instructions and calling list. When the calling has been completed, the email address in your config file will be sent both CSV and XML reports of the results of all the calls. Either of these reports is suitable for import and manipulation using most spreadsheet applications.

Status Codes. Every call that is processed gets a status code entry whether the call is successful or fails. A status code of 0 means a call failed to both phone numbers provided for a particular callee. The second phone number is entirely optional. A status code of 5 means the call was answered but no response was provided by the called party. This typically would mean the call was picked up by an answering machine although it could mean Granny answered the call using a rotary dial phone. 🙄 Status codes of 1 through 4 have whatever meaning you choose to assign to each option when setting up a configuration for a particular calling campaign.

Legalese. TeleYapper 5.0 is free for use by non-profit, civic, and non-political organizations provided you absolve us from all financial and other responsibility in conjunction with your use of the software. Non-profit use further requires that no financial benefit be derived from the substance of the calls. Simply stated, your Little League team can use the software at no cost to remind kids to attend practice, but it cannot be used to solicit charitable contributions or to sell doughnuts without obtaining a commercial license.

By using this software, you also agree to strictly comply with federal and state regulations including 16 C.F.R. Part 310. In addition, you agree to assume all risks associated with use of the software. NO WARRANTIES EXPRESS OR IMPLIED INCLUDING ITS FITNESS FOR USE OR MERCHANTABILITY ARE PROVIDED WITH THIS SOFTWARE.

WARNING: With certain limited exceptions, most robocalling now requires prior written approval from those being called. See this link for a summary of the federal requirements. Be advised that improper use of this software may subject the user to penalties of up to $16,000 per call plus monetary damages to injured consumers.

Creative Commons LicenseLicensing. You are licensed to use this software under certain conditions. You do not own it. We do, and we also own the copyright. It is licensed for use under the terms of the Creative Commons Attribution Non-Commercial license. A Plain English summary is available here. We've done this primarily to do our part to stamp out the telemarketing creeps of the world. Those wishing to use TeleYapper for commercial or political purposes must first request and then purchase a commercial license after outlining your proposed terms of use. Telemarketers need not apply! For doctors, lawyers, and others falling outside the scope of our free license who wish to obtain a commercial use license, please contact us for pricing and details. Be sure to summarize your intended use in your request together with a sufficient factual summary to demonstrate that your use is in compliance with 16 C.F.R. Part 310. Please also indicate whether you will require assistance with installation and setup.

Prerequisites. As mentioned, you'll need a Linux-based Asterisk aggregation such as PBX in a Flash to use TeleYapper 5.0. This means you need a system with Asterisk 1.4 or 1.6 as well as FreePBX 2.5 or higher. For quality reasons, we strongly recommend you purchase a commercial Cepstral text-to-speech license for your server. While Flite would technically work, most folks don't respond well to calls from Egor so we have customized the code for use solely with Cepstral. You'll find Cepstral installation instructions in this Nerd Vittles article. The TeleYapper 5.0 code also relies heavily on Apache and PHP, both of which are included in every PBX in a Flash system.

Installing Cepstral. Cepstral installation is not the simplest application to get working with Asterisk so here are the commands for those running 32-bit systems with Asterisk 1.4 or 1.6.2. For details on purchasing and registering Cepstral (and a discount) and for 64-bit installs, read our previous article including the comments.

For Asterisk 1.4 systems running under 32-bit CentOS, log into your server as root and issue the following commands accepting the Cepstral defaults. Be sure to create the Cepstral directory when prompted!

cd /root
wget http://nerd.bz/bnTVjX
tar -zxvf Cepstral*
cd Cepstral_Allison-8kHz_i386-linux_5.1.0
./install.sh
echo /opt/swift/lib > /etc/ld.so.conf.d/cepstral.conf
ldconfig
cd /usr/src
wget http://pbxinaflash.net/source/app_swift/app_swift-1.4.2.tar.gz
tar -zxvf app_swift*
cd app_swift-1.4.2
make
make install
ln -s /opt/swift/bin/swift /usr/bin/swift
sed -i 's|David-8kHz|Allison-8kHz|' /etc/asterisk/swift.conf
amportal restart
asterisk -rx "core show application swift"
ls /opt/swift/voices
swift --reg-voice

For Asterisk 1.6.2 systems running under 32-bit CentOS, log into your server as root and issue the following commands accepting the Cepstral defaults. Be sure to create the Cepstral directory when prompted!

cd /root
wget http://nerd.bz/bnTVjX
tar -zxvf Cepstral*
cd Cepstral_Allison-8kHz_i386-linux_5.1.0
./install.sh
echo /opt/swift/lib > /etc/ld.so.conf.d/cepstral.conf
ldconfig
cd /usr/src
wget http://pbxinaflash.net/source/app_swift/app_swift-1.6.2.tar.gz
tar -zxvf app_swift*
cd app_swift-1.6.2
make
make install
ln -s /opt/swift/bin/swift /usr/bin/swift
sed -i 's|David-8kHz|Allison-8kHz|' /etc/asterisk/swift.conf
amportal restart
asterisk -rx "core show application swift"
ls /opt/swift/voices
swift --reg-voice

Installing TeleYapper 5.0 The real beauty of PBX in a Flash as an Asterisk platform is demonstrated by the ease with which you can install new applications such as this one. The drill is very simple. You download an install script, make it executable, and run it. Less than a minute later, the TeleYapper install is done. Here are the commands to execute to install TeleYapper 5.0 after logging into your PBX in a Flash system as root. On other systems, you are well advised to carefully review the install script and tailor it to meet the individual requirements of the platform on which you are installing it.

cd /root
wget http://bestof.nerdvittles.com/applications/teleyapper5/teleyapper5.pbx
chmod +x teleyapper5.pbx
./teleyapper5.pbx

The TeleYapper Database. We use the MySQL database management system to manage the list of callees for TeleYapper to dial. It can handle a database of almost any size and generally stands up well in performance comparisons with Oracle. So you're covered on the database front. For most users, you never should need to access the MySQL database directly. TeleYapper 5.0 handles the importing of CSV or XML files for processing, manages the call queue, and processes and emails CSV and/or XML-formatted reports to you when the calls are completed.

The install script creates the MySQL database to support TeleYapper 5.0. Should you need or want to manage the database directly, the easiest tool to use is phpMyAdmin which is accessible through the Tools tab in FreePBX on PBX in a Flash systems. You'll need to login as maint with your maint password to access phpMyAdmin. After phpMyAdmin loads, click on the reminders database in the left column. Then click the reminders table entry in the left column to open the file. Unless you really, really know what you are doing and appreciate how much coding will be required to support new or different fields in the reminders file, don't improve it.

Here's the layout of the MySQL database table for TeleYapper 5.0:

  • id - System generated record ID
  • acctno - Account Number (12 alphanumeric characters)
  • provider - Provider Name (30 alphanumeric characters)
  • recipient - Recipient Name (30 alphanumeric characters)
  • apptdt - Appointment Date (MM/DD/YY format)
  • appttime - Appointment Time (HHMM format using 24-hr clock)
  • apptplace - Appointment Location (30 alphanumeric characters)
  • instructions - Free-form text (65535 alphanumeric characters)
  • phone1 - Primary Phone (NNN-NNN-NNNN or NNNNNNNNNN)
  • phone2 - Alternate Phone (NNN-NNN-NNNN or NNNNNNNNNN)
  • status - Status: 0=failedcall 5=ansmachine 1,2,3,4=user-defined
  • failedcalls - System Generated Number of Failed Calls

Tweaking PHP for TeleYapper. Depending upon your PHP setup and the number of calls you plan to process, you may need to adjust the default PHP resource settings on your server. The main reason is because TeleYapper generates a custom sound file for every call to be processed before the calling ever starts. If you plan to make thousands of calls, this can take some time. The PHP settings are stored in /etc/php.ini. You must log in as root and restart Apache after making changes to these settings: service httpd restart. The settings that matter are the following:

max_execution_time = 30 (we recommend 900 which is 15 minutes to process)
max_input_time = 60 (we recommend 300 which is 5 minutes to upload a file)
memory_limit = 100M (OK as is)

post_max_size = 8M (we recommend 100 megabytes which should be ample)

file_uploads = On (OK as is on most systems)
upload_max_filesize = 100M (we recommend 100 megabytes which should be ample)

Tweaking Crontab. TeleYapper relies upon a cron job to kick off its calling sprees so you'll need the following entry in your /etc/crontab file unless you used the install script which inserts it automatically:

* * * * * root /var/www/html/appt-reminders/gen-reminders.php > /dev/null 2>&1

Formatting CSV Data For Import. You don't necessarily need an external database in order to use TeleYapper 5.0 although it is designed to support almost any database or spreadsheet application in the marketplace so long as it can export data in CSV or XML format. A CSV (comma-separated values) or XML file is the middleware that makes everything work. Each line in a CSV file represents an entry to be processed by TeleYapper 5.0 when the CSV file is uploaded. Each item in a line is called a field. Every field begins and ends with double-quotes, and fields are separated from each other with commas. Do NOT include any quotation marks in your actual text, or you'll get a disaster. All fields are required, by the way, but only the Phone1 field must have an actual entry. The remaining fields may each consist of nothing more than a pair of double-quotes. Note also that the id, status, and failedcalls fields (shown in red below) must consist of a pair of double-quotes and nothing more. Here's the actual CSV format which must be used, and all of the data must appear on the same line so disregard the WordPress formatting below:

"id","acctno","provider","recipient","apptdt","appttime","apptplace","instructions","phone1","phone2",
"status","failedcalls"

Here's what the CSV entry used for our sample medical reminder shown near the top of this article would look like. We've excluded the special instructions and Phone2 entries below only to simplify the display because of constraints inherent in our blog formatting:

"","12345","Quack","Jan","10/05/10","1030","Charleston","","4049876543","","",""

The XML Alternative. If you'd prefer to upload XML file templates for your calls instead of CSV data, a sample XML file is included in the distribution to show you the proper formatting. Here's a sample entry that matches the CSV data above:

<!-- Database: reminders -->
<reminders>
   <!-- Table: reminders -->
    <reminders>
       <id></id>
       <acctno>12345</acctno>
       <provider>Quack</provider>
       <recipient>Jan</recipient>
       <apptdt>10/05/10</apptdt>
       <appttime>1030</appttime>
       <apptplace>Charleston</apptplace>
       <instructions></instructions>
       <phone1>4049876543</phone1>
       <phone2></phone2>
       <status></status>
       <failedcalls></failedcalls>
    </reminders>
</reminders>

Direct Uploading with SAMBA. If you've activated SAMBA on your Asterisk server, you can upload TeleYapper files for processing directly. Be sure to name your CSV or XML file as reminders.csv or reminders.xml. And name your config file: config.php. Copy the files to the /var/www/html/appt-reminders/upload directory on your Asterisk server. That's all there is to it. If you need hints on SAMBA installation, see our Best of Nerd Vittles tutorial. Pay particular attention to the sections on Security Considerations and Firewall Settings. Before using the SAMBA, be sure to upload some test CSV/XML files using the web interface. There is no error checking when you use the SAMBA option!

Configuring TeleYapper 5.0 Calling Scripts. Now let's address how we transform a CSV or XML entry such as the ones shown above into a personalized phone call to Jan, the actual patient in our example. Every TeleYapper session can have an individual configuration file associated with it. If none is specified, then a default configuration is used. In this way, you can customize call procedures and calling scripts for different tasks. The easiest approach is to always upload a config file with your CSV or XML data file. Then you won't get unexpected results when the calling begins.

HINT: It's a very good idea to create a sample upload with your own phone number and some sample configuration data to test things out before you start calling thousands of clients.

A default configuration file (config.default.php) as well as sample CSV and XML templates (reminders.csv and reminders.xml) come with TeleYapper 5.0 and can be found on your Asterisk server in the /var/www/html/appt-reminders directory. Make a copy of them, and move the copies to your Mac or PC. Then, using TextEdit or Notepad, open the files and have a look. Before addressing other configuration options in config.php, let's tackle the setup procedure for calling scripts.

The actual boilerplate message to be delivered to the called party is stored in $msg. Notice that you can substitute data out of your database in the boilerplate template by enclosing any desired fields in braces. Just make sure the fieldname exactly matches one of the fields in the reminders database. So our entry for the sample call above would look like this:

$msg="Hi: This is Allison from Charleston Family Clinic calling to confirm an appointment for {recipient}, with Doctor {provider}, on {apptdt}, at {appttime}, in our {apptplace} office. {instructions}";

Just a comment that, for those with large data processing systems, you may find it more convenient to generate the actual text for each reminder on your mega-machine. In this case, all of the data (up to 65,535 characters) could be loaded into the instructions field for each callee. So each upload record might consist of nothing more than phone numbers and instructions. In this scenario, the $msg entry in config.php would look like this: $msg="{instructions}";

The key press choices that are provided to the called party are configured using the $options field which would look like this for our example:

$options = "To confirm your appointment, please press 1. To reschedule your appointment, press 2. To cancel your appointment, press 3. If we have reached you in error or if you do not wish to receive appointment reminders, press 4. To hear this message again, please press 5 now.";

Don't confuse the 5 option which is automatically included in the TeleYapper dialplan code with status code 5 which means an answering machine picked up a call. Status code 5 is system-generated and is not stored based upon a callee choosing to listen to a recorded message more than once. The two 5's are not the same even though options 1-4 are actually used to define what the first four status codes mean on your system.

As we mentioned, the system has the smarts to usually figure out if an answering machine took the call. When it detects this, the $ansmach message is played instead of $options. A sample entry might look like this:

$ansmach = "If you need to cancel or reschedule this appointment, if we reached you in error, or if you do not wish to receive appointment reminders in the future, please call 777-123-4567 at your earliest convenience. Thank you for your assistance. Goodbye.";

Finally, for each of the four choices (1 through 4), there is a response message which is played if the callee chooses that option. Here's a sample template to get you started:

$chose1 = "Thank you for making Charleston Family Clinic your medical home. Your appointment has been confirmed. Goodbye.";
$chose2 = "Thank you. A representative will be calling you to reschedule your appointment. Goodbye.";
$chose3 = "Thank you for making Charleston Family Clinic your medical home. Your appointment has been cancelled. Goodbye.";
$chose4 = "Thank you. We will update our systems and apologize for the call. Goodbye.";

Thus, when a callee responds to the boilerplate call by pressing 1, $chose1 is played in response. If an email address has been entered for $chose1email, then a copy of the log entry for that call is sent to the specified email address using the customized email subjects (shown below) in addition to being placed in the master call log. The same process occurs when the other options are chosen. Particularly with medical appointment cancellations, it may be important to receive immediate notification when an appointment is canceled or a patient requests a change in scheduling. So the software includes the flexibility to generate instant emails to various email addresses depending upon which option is pressed. As noted, the optional instant emails will be generated using the email subjects entered for the following fields in your customized configuration file:

$chose1subj = "APPOINTMENT NOTIFICATION CONFIRMED BY PHONE";
$chose2subj = "APPOINTMENT RESCHEDULING REQUEST BY PHONE";
$chose3subj = "APPOINTMENT CANCELLATION REQUEST BY PHONE";
$chose4subj = "APPOINTMENT SCHEDULING ERROR REPORTED BY PHONE";
$chose5subj = "APPOINTMENT NOTIFICATION LEFT ON ANSWERING MACHINE";

Uploading Data & Config Files to TeleYapper. Simple web pages are used to upload CSV and XML data with config files to TeleYapper 5.0. WARNING: These web pages have NOT been sanitized for use on the Internet. They are designed for use on your local area network behind a secure firewall. On PBX in a Flash systems, the web pages are password-protected and require a valid user account login for access. This will NOT be the case on other Asterisk aggregations without tweaking your Apache configuration. Sample entries can be found in teleyapper.conf in the /var/www/html/appt-reminders directory. On PBX in a Flash systems, you can log in using maint, wwwadmin, or meetme accounts. Or you can create an additional account to use with TeleYapper 5.0:

htpasswd /usr/local/apache/passwd/wwwpasswd teleyapper

There are separate web pages depending upon whether you wish to upload CSV or XML data. For CSV data, the web address is http://ipaddress/appt-reminders/uploadcsv/. For XML data, the web address is http://ipaddress/appt-reminders/uploadxml/. Substitute the private IP address of your Asterisk server for ipaddress. Here's a sample of the CSV web form. You can, of course, substitute your own logo on the right if desired.

CSV Web Form

Other TeleYapper 5.0 Config Options. In addition to the boilerplate text for TeleYapper calls, there are a number of other settings which can be adjusted to meet your individual requirements.

The database settings should never need adjusting so just leave them alone. They look like this:

$db="reminders";
$fi="reminders";
$dbuser="root";
$dbpass="passw0rd";

You can manually set a starting and ending time to begin and end the calling sequence for a particular upload. Never set these in the default configuration! Only set them in a config file to be uploaded. If the entries are blank, calls will commence shortly after the upload completes and will end when all of the entries have been processed. Note that there is no current flexibility to schedule individual calls based upon the time of the appointment. This typically would be handled by selecting particular records for processing in your primary database. For example, for medical appointments, you would select records in which an appointment is scheduled for tomorrow and then upload the list to TeleYapper which would place the calls today. We probably will expand this functionality down the road, but it's not there yet. So it's up to you to upload call lists which basically are ripe for calling now.

If you wish to use the $startcalls and $endcalls features in your custom config files, the syntax should look like this: YYYYMMDD,HHMM where YYYY is a 4-digit year, MM is a 2-digit month, DD is a 2-digit day of the month, HH is the 2-digit hour based upon a 24-hour clock (aka Military Time), and MM is the 2-digit minute. Note that calls will not end precisely at the $endcalls time. Any existing calls already in process will be completed including redials and calls to an alternate $phone2 number. This process can take up to 10 minutes to complete.

CAUTION: Be very careful using the $startcalls option! Nothing precludes your scheduling a thousand reminder calls to kick off at 0200 which is 2 a.m. Not really a good thing if job security matters to you.

To restart the calling process on the following day, log into your server as root and switch to the /var/www/html/appt-reminders directory. Then edit config.php and adjust the $startcalls and $endcalls for the remaining calls. Then run: ./gen-calls.php. Any existing database entry with a status=0 will be called when the calling process resumes. You can monitor the calling process by running: ./showcalls.sh. Press Ctrl-C to terminate the call display. It usually takes a minute or two for the first call to be placed.

$callerid is used to set the CallerID of outbound calls if your telephony provider supports it.

$trunk is used to set the outbound dialing trunk for calls. The default works for most purposes.

$channel is used to set the outbound dialing channel for calls. The default works for most purposes.

$maxcalls and $spacing are used to set the number of simultaneous calls and spacing between calls respectively. Be very careful with these settings. You must have sufficient outbound trunks to handle the number of simultaneous calls you schedule with $maxcalls, or you will get circuit busy conditions which are recorded as calls to busy numbers. Keep in mind that TeleYapper tries every call twice with 2 minutes of separation. So, if you only have two outbound trunks, don't set $maxcalls above 1, or you will get trunk busy conditions whenever original calls to an individual fail, i.e. line busy or no answer situations. In addition, remember that TeleYapper 5.0 supports a second phone number for each called party. These are triggered whenever the original two calls to the primary number fail and must also be considered in setting $maxcalls properly. If your logs show a disproportionate number of failed calls (status=0), this may be a tell-tale sign of trunk busy conditions.

$waittime is the number of seconds a call to any given number will ring. 45 seconds is about 7 rings.

$email is the email address that will be used to send the logs at the completion of the calling process. $chose1email through $chose5email are the optional email addresses if you want instantaneous feedback on certain types of status results. This means you get an immediate email if a certain call results in a certain status code. Leave the ones blank for $status conditions on which you want no immediate feedback and simply wait for the logs to arrive.

$csvreport and $xmlreport are used to set which type of completion report you wish to receive. If you want both of them, set them both to 1. Otherwise, set the one you don't want to 0.

The Old Fashioned Way. For those of you that preferred the older method of entering data directly into MySQL, you still can use phpMyAdmin or some other front-end tool to enter the data directly into the reminders.reminders table. Just leave the id field blank since it automatically gets generated by MySQL. And either leave the status and failedcalls fields blank or set them to 0. They also are system-generated. Once you have your data in place, log into your server as root, and...

cd /var/www/html/appt-reminders
Configure config.php for your calling campaign
Run ./gen-mysql.php to kick off TeleYapper 5.0

In Closing... Finally, let us issue our usual tinkerer's warning. Don't delete anything from the /var/www/html/appt-reminders directory tree. Just because you don't know its function doesn't mean it doesn't have one. Aside from that, the documentation above should get you started today. Be advised that TeleYapper 5.0 still is a work in progress. So check back every week or so for new comments on this article to see what's been changed, added, or fixed since you originally downloaded the application. Enjoy!


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